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TRANSCRIPT
IxLoad Voice Test Library Reference
Release 8.00December 2015
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ii IxLoad Voice Test Library Reference Guide, Release 8.00
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Table of Contents
Chapter 1 Introduction to the VoiceTest Libraries
Who Should Read This Manual. . . . . . . . . . . . . . . . . . . . . . . 1-1
How This Manual Is Organized . . . . . . . . . . . . . . . . . . . . . . . 1-1
Voice Test Libraries Overview . . . . . . . . . . . . . . . . . . . . . . . . 1-2
Voice Libraries Function Sets . . . . . . . . . . . . . . . . . . . . . . . . 1-2SIP Library Functions Set . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-2Skinny Library Functions Set. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-3RTP Library Functions Set . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-4Flow Functions Set . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-5MGCP Functions Set. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-6T1/E1 Functions Set . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-7
Chapter 2 Voice Test Libraries Settings
Global Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1Configuring Library Settings and Outputs . . . . . . . . . . . . . . . . . . . . 2-1
RTP Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2-2
Skinny Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2-3
SIP Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2-5
STUN Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2-6
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T.38 Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-6
PSTN Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-6
Function Outputs Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-10
Configuring Scenario Editor Default Settings . . . . . . . . . . . . . . . . .2-12
Chapter 3 Voice Functions Reference
VoIP SIP Functions Library . . . . . . . . . . . . . . . . . . . . . . . . . 3-1Send Request . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-2
Send Request Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-2
Send Request Properties: Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-4
Send Request Properties: Flow Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-5
Send Request Properties: Extract Variables . . . . . . . . . . . . . . . . . . . . . . . . . . 3-8
Send Request Properties: Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-8
Send Request Properties: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-9
Send Response . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-9Send Response Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-9
Send Response Properties: Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-11
Send Response Properties: Flow Manager. . . . . . . . . . . . . . . . . . . . . . . . . . 3-12
Send Response Properties: Extract Variables . . . . . . . . . . . . . . . . . . . . . . . 3-13
Send Response Properties: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-13
Wait Request. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-13Wait Request Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-13
Wait Request Properties: Extract Variables . . . . . . . . . . . . . . . . . . . . . . . . . 3-14
Wait Request Properties: Retransmission. . . . . . . . . . . . . . . . . . . . . . . . . . . 3-14
Wait Request Properties: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-15
Wait Response . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-16Wait Response Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-16
Wait Response Properties: Extract Variables . . . . . . . . . . . . . . . . . . . . . . . . 3-17
Wait Response Properties: Retransmission . . . . . . . . . . . . . . . . . . . . . . . . . 3-17
Wait Response Properties: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-19
Wait Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-19Wait Message Properties: Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-19
The Template Window. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-20
Message Header Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-21
Wait Message Properties: Params . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-24
Wait Message Properties: Extract Variables . . . . . . . . . . . . . . . . . . . . . . . . . 3-25
Wait Message Properties: Retransmission . . . . . . . . . . . . . . . . . . . . . . . . . . 3-25
Wait Message Properties: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-26
Retransmit Last Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-27Retransmit Last Message: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-27
Extract Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-27
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Extract Variables: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-27
Variable Operations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-27
Extraction Rules Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-28
Extract Variables: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-32
MSRP Send AUTH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-32MSRP Send Auth: Tx Request Parameters . . . . . . . . . . . . . . . . . . . . . . . . . .3-32
MSRP Send Auth: Rx Response Parameters. . . . . . . . . . . . . . . . . . . . . . . . .3-32
MSRP Send Auth: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-33
MSRP Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-33MSRP Session: Content Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-33
MSRP Session: Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-34
MSRP Session: Tx Requests Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . .3-35
MSRP Session: Tx Responses Parameters . . . . . . . . . . . . . . . . . . . . . . . . . .3-36
MSRP Session: CPIM Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-36
MSRP Session: Output Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-38
MSRP Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-38MSRP Control: Control Action Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . .3-39
MSRP Control: Output Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-39
VoIP Skinny Functions Library . . . . . . . . . . . . . . . . . . . . . . 3-41RegisterClient . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-41UnregisterClient. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-45OffHook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-46OnHook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-48NewCall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-49EndCall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-50MakeCall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-51WaitCall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-52AnswerCall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-53DialDigits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-55WaitDigits. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-56HoldCall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-57RetrieveCall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-58Setup XFER . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-59Complete XFER. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-61Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-63ForwardAllCalls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-66ParkCall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-67GetCallInfo. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-68MeetMe . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-68
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RemoveLastConferenceParty . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-70
SendStimulus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-71SendSoftkey . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-72IsSoftKeyAvailable . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-73WaitForEvent . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-74
VoIP Media Functions Library . . . . . . . . . . . . . . . . . . . . . . 3-86Generate DTMF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-86
Generate DTMF: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-86
Generate DTMF: Advanced Playback Settings . . . . . . . . . . . . . . . . . . . . . . . 3-87
Generate DTMF: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-88
Detect DTMF. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-88Detect DTMF: Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-88
Detect DTMF: Advanced Detection Settings . . . . . . . . . . . . . . . . . . . . . . . . . 3-89
Detect DTMF: Outputs. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-90
Generate MF. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-90Generate MF: Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-91
Generate MF: Advanced Playback Settings . . . . . . . . . . . . . . . . . . . . . . . . . 3-91
Generate MF: Outputs. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-91
Detect MF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-92Generate Tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-92
Generate Tone: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-92
Generate Tone: Advanced Playback Settings. . . . . . . . . . . . . . . . . . . . . . . . 3-92
Generate Tone: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-93
Wait for Tone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-93Wait for Tone: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-93
Wait for Tone: Advanced Detection Settings. . . . . . . . . . . . . . . . . . . . . . . . . 3-93
Wait for Tone: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-94
Talk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-94Talk: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-94
Talk: Advanced Playback Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-95
Talk: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-95
Listen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-95Listen: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-96
Listen: Advanced Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-96
Listen: Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-97
Voice Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-97Voice Session: Talk Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-97
Voice Session: Listen Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-97
Voice Session: Advanced Playback Settings . . . . . . . . . . . . . . . . . . . . . . . . 3-97
Voice Session: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-98
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Multimedia Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-98Multimedia Session: Video Play Parameters . . . . . . . . . . . . . . . . . . . . . . . . .3-98
Multimedia Session: Advanced Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-99
Multimedia Session: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-100
T.38 Fax Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-100T.38 Fax Session: General Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-100
T.38 Fax Session: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-102
Path Confirmation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-102Path Confirmation: Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-103
Path Confirmation: Tone Detection/Generation Page. . . . . . . . . . . . . . . . . .3-105
Path Confirmation: Advanced Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-105
Path Confirmation: Output Settings. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-106
RTP Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-107RTP Control: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-107
RTP Control: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-108
Forceful RTP Function Termination and Statistics . . . . . . . . . . . . . . . . .3-109
Warnings and Errors. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-110
VoIP Flow Functions Library . . . . . . . . . . . . . . . . . . . . . . . 3-111Start . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-111
Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-111
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-111
Stop . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-111Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-111
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-111
Variable Set . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-111Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-111
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-112
Variable Test . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-112Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-112
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-113
Sleep . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-113Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-113
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-113
Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-114Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-114
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-114
Exit Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-114Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-114
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-114
Counter Op . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-114Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-114
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-114
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Test Time. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-115Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-115
Log Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-115Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-115
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-115
Dump Variables. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-115Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-115
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-116
Error Handler . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-116Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-116
Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-116
VoIP H323 RAS Library . . . . . . . . . . . . . . . . . . . . . . . . . . 3-116H323 Register . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-116
Register Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-117
H323 UnRegister . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-117H.323 Unregister: Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-118
H323 Unregister: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-118
VoIP H323 Functions Library . . . . . . . . . . . . . . . . . . . . . . 3-118Make Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-118
Make Call Properties: Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-118
Make Call Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-119
Receive Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-120Receive Call Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-120
Receive Call Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . 3-121
End Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-122End Call Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-122
VoIP H248 Functions Library . . . . . . . . . . . . . . . . . . . . . . 3-123H.248 MGC Library . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-123Add . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-123
Add: Tx Request . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-123
Add: Rx Reply . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-125
Add: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-126
Add: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-127
Modify . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-127Move . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-127Subtract. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-128AuditVal. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-128AuditCap . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-128
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SrvChange (MGC). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-128Wait Notify . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-128
Wait Notify: Rx Request . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-128
Wait Notify: Tx Reply. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-130
Wait Notify: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-132
Wait Notify: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-133
Wait SrvChange (MGC). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-133Wait Requests (MGC) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-134H.248 MGW Library. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-135Notify . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-135SrvChange (MGW) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-135Wait Add . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-135Wait Modify . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-136Wait Move . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-136Wait Subtract . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-136Wait AuditVal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-136Wait AuditCap . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-137Wait SrvChange (MGW) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-137Wait Requests (MGW). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-137
VoIP MGCP Functions Library . . . . . . . . . . . . . . . . . . . . . 3-138Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-138MGCP MGW Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-138
Send NTFY . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-139
Send Notify Properties:Tx Command . . . . . . . . . . . . . . . . . . . . . . . . . . .3-139
Send Notify Properties:Rx Response . . . . . . . . . . . . . . . . . . . . . . . . . . .3-139
Send Notify Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-139
Send Notify Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . .3-139
Send DLCX (GW) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-140
Send DLCX Properties:Tx Command. . . . . . . . . . . . . . . . . . . . . . . . . . .3-140
Send Notify Properties:Rx Response . . . . . . . . . . . . . . . . . . . . . . . . . . .3-140
Send DLCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-140
Send DLCX Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . .3-140
Send RSIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-141
Send RSIP Properties:Tx Command . . . . . . . . . . . . . . . . . . . . . . . . . . .3-141
Send RSIP Properties:Rx Response . . . . . . . . . . . . . . . . . . . . . . . . . . .3-141
Send RSIP Properties:Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-141
Send RSIP Properties:Output Settings. . . . . . . . . . . . . . . . . . . . . . . . . .3-141
Wait CRCX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-141
Wait CRCX Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . .3-141
Wait CRCX Properties:Tx Response . . . . . . . . . . . . . . . . . . . . . . . . . . .3-141
Wait CRCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-141
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Wait CRCX Properties:Output Settings. . . . . . . . . . . . . . . . . . . . . . . . . 3-141
Wait DLCX (GW) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait DLCX Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait DLCX Properties:Tx Response. . . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait DLCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait DLCX Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait MDCX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait MDCX Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait MDCX Properties:Tx Response . . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait MDCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-142
Wait MDCX Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait RQNT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait RQNT Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait RQNT Properties:Tx Response . . . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait RQNT Properties:Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait RQNT Properties:Output Settings. . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait AUEP. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait AUEP Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait AUEP Properties:Tx Response. . . . . . . . . . . . . . . . . . . . . . . . . . . 3-143
Wait AUEP Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait AUEP Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait AUCX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait AUCX Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait AUCX Properties:Tx Response . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait AUCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait AUCX Properties:Output Settings. . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait EPCF. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait EPCF Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144
Wait EPCF Properties:Tx Response. . . . . . . . . . . . . . . . . . . . . . . . . . . 3-145
Wait EPCF Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-145
Wait EPCF Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . 3-145
Wait Any Command (GW) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-145
Wait Any Command (GW) Properties:Rx Command . . . . . . . . . . . . . . 3-145
Wait Any Command (GW) Properties:Tx Response . . . . . . . . . . . . . . . 3-145
Wait Any Command (GW) Properties: Parameters. . . . . . . . . . . . . . . . 3-145
Wait Any Command (GW) Properties: Output Settings . . . . . . . . . . . . 3-146
Wait Command (GW) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-146
Wait Command Properties: Templates . . . . . . . . . . . . . . . . . . . . . . . . . 3-146
Wait Command Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . 3-147
Wait Command Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . 3-148
MGCP CA Functions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-148Send RQNT. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-148
Send RQNT Properties:Tx Command. . . . . . . . . . . . . . . . . . . . . . . . . . 3-148
Send RQNT Properties:Rx Response. . . . . . . . . . . . . . . . . . . . . . . . . . 3-149
Send RQNT Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-149
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Send RQNT Properties:Output Settings. . . . . . . . . . . . . . . . . . . . . . . . .3-149
Send CRCX. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-149
Send CRCX Properties:Tx Command . . . . . . . . . . . . . . . . . . . . . . . . . .3-149
Send CRCX Properties:Rx Response . . . . . . . . . . . . . . . . . . . . . . . . . .3-149
Send CRCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-149
Send CRCX Properties:Output Settings. . . . . . . . . . . . . . . . . . . . . . . . .3-149
Send DLCX (CA). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-149
Send DLCX Properties:Tx Command. . . . . . . . . . . . . . . . . . . . . . . . . . .3-149
Send DLCX Properties:Rx Response. . . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send DLCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send DLCX Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send MDCX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send MDCX Properties:Tx Command . . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send MDCX Properties:Rx Response . . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send MDCX Properties:Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send MDCX Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send AUCX. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send AUCX Properties:Tx Command . . . . . . . . . . . . . . . . . . . . . . . . . .3-150
Send AUCX Properties:Rx Response . . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send AUCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send AUCX Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send AUEP. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send AUEP Properties:Tx Command . . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send AUEP Properties:Rx Response . . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send AUEP Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send AUEP Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send EPCF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send EPCF Properties:Tx Command. . . . . . . . . . . . . . . . . . . . . . . . . . .3-151
Send EPCF Properties:Rx Response. . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Send EPCF Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Send EPCF Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Wait NTFY. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Wait NTFY Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Wait NTFY Properties:Tx Response. . . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Wait NTFY Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Wait NTFY Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Wait DLCX (CA) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Wait DLCX Properties:Rx Command . . . . . . . . . . . . . . . . . . . . . . . . . . .3-152
Wait DLCX Properties:Tx Response . . . . . . . . . . . . . . . . . . . . . . . . . . .3-153
Wait DLCX Properties:Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-153
Wait DLCX Properties:Output Settings. . . . . . . . . . . . . . . . . . . . . . . . . .3-153
Wait Command (CA) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-153
Wait Any Command (CA) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-153
Wait RSIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-153
Wait RSIP Properties:Rx Command. . . . . . . . . . . . . . . . . . . . . . . . . . . .3-153
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Wait RSIP Properties:Tx Response . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-153
Wait RSIP Properties:Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-154
Wait RSIP Properties:Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . 3-154
Digital T1/E1 Functions Library . . . . . . . . . . . . . . . . . . . . 3-155Make Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-155
Make Call Properties: Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-155
Make Call Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-156
Receive Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-157Receive Call Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-157
Receive Call Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . 3-158
End Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-158End Call Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-158
Path Confirmation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-159Path Confirmation Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . 3-159
Path Confirmation Properties: Tone Detection /Generation . . . . . . . . . . . . 3-161
Path Confirmation Properties: Advanced Settings . . . . . . . . . . . . . . . . . . . 3-162
Path Confirmation Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . 3-163
Talk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-163Talk Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-163
Talk Properties: Advanced Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-164
Talk Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-164
Listen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-165Listen Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-165
Listen Properties: Output Settings. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-166
Voice Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-166Voice Session Properties: Talk Parameters . . . . . . . . . . . . . . . . . . . . . . . . 3-167
Generate DTMF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-167Generate DTMF Properties: Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . 3-167
Generate DTMF Properties: Advanced Settings . . . . . . . . . . . . . . . . . . . . . 3-168
Generate DTMF Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . . 3-169
Detect DTMF. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-169Detect DTMF Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-169
Detect DTMF Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . . 3-170
Generate MF. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-171Detect MF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-171Generate Tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-171
Generate Tone Properties: Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-171
Generate Tone Properties: Advanced Settings . . . . . . . . . . . . . . . . . . . . . . 3-172
GenerateTone Properties: Output Settings . . . . . . . . . . . . . . . . . . . . . . . . . 3-172
Wait for Tone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3-173Wait for Tone Properties: Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-173
Wait for Tone Properties: Output Settings. . . . . . . . . . . . . . . . . . . . . . . . . . 3-174
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Chapter 4 Basic Test Scenarios and Procedures
SIP Procedures, Sample Test Configurations and Test Scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-1
SIP Predefined Procedures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-2VoIPSIP Peer
Test Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-6VS_001_B2B_SIPv4 MakeCall - ReceiveCall - EndCall. . . . . . . . . . . . . . . . . .4-7
VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s . . . . .4-7
VS_003_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with SRTP - 33s . . . .4-9
VS_004_B2B_SIPv4 MakeCall - ReceiveCall - EndCall - Early Media. . . . . . .4-9
VS_005_DUT_SIPv4 MakeCall - ReceiveCall - EndCall with RecordRoute. .4-11
VS_006_DUT_SIPv4 MakeCall - ReceiveCall with Registration . . . . . . . . . .4-12
VS_007_DUT_SIPv4 Make - Receive Call with ReRegistration . . . . . . . . . . .4-14
VS_008_DUT_SIPv4 MakeCall - Receive Call with Registration - Complete.4-15
VS_009_B2B_SIPv4 MakeCall - ReceiveCall with Tel URI - Global Phone Num-bers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-16
VS_010_B2B_SIPv4 Make - Receive Call with Tel URI - Local Phone Numbers 4-16
VS_011_B2B_SIPv4 Basic Transfer - Successful . . . . . . . . . . . . . . . . . . . . .4-19
VS_012_B2B_SIPv4 Basic Transfer - Target Busy . . . . . . . . . . . . . . . . . . . .4-20
VS_013_B2B_SIPv4 Basic Transfer - Target No Answer. . . . . . . . . . . . . . . .4-20
VS_014_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with Hold UnHold . .4-21
VS_015_DUT_SIPv4 Hold - UnHold with Registration and Path Confirmation . . 4-22
VS_016_DUT_SIPv4 Send - Receive MESSAGE with Registration . . . . . . .4-24
VS_017_DUT_SIPv4 IMS MakeCall - ReceiveCall with Registration and RTP . . 4-25
VS_018_DUT_SIPv4 IMS Registration with Subscription. . . . . . . . . . . . . . . .4-27
VS_019_DUT_SIPv4 MakeCall - ReceiveCall with RTP - SBC Testing . . . . .4-29
VS_020_DUT_SIPv4 ReceiveCall - MakeCall with RTP - SBC Testing . . . . .4-31
VS_021_DUT_SIPv4 MakeCall - EndCall. . . . . . . . . . . . . . . . . . . . . . . . . . . .4-34
VS_022_DUT_SIPv4 MakeCall - EndCall with RTP - 33s . . . . . . . . . . . . . . .4-35
VS_023_DUT_SIPv4 MakeCall - EndCall with Hold Unhold . . . . . . . . . . . . .4-36
VS_024_DUT_SIPv4 MakeCall - EndCall with SRTP - 33s . . . . . . . . . . . . . .4-36
VS_025_DUT_SIPv4 MakeCall - EndCall through SIP Redirect Server. . . . .4-36
VS_026_DUT_SIPv4 ReceiveCall - EndCall . . . . . . . . . . . . . . . . . . . . . . . . .4-37
VS_027_DUT_SIPv4 ReceiveCall - EndCall with RTP - 33s . . . . . . . . . . . . .4-38
VS_028_DUT_SIPv4 ReceiveCall - EndCall with SRTP - 33s . . . . . . . . . . . .4-40
VS_029_DUT_SIPv4 ReceiveCall - EndCall with Hold Unhold . . . . . . . . . . .4-40
VS_030_B2B_SIPv4_TLS_MakeCall - ReceiveCall - EndCall . . . . . . . . . . . .4-40
VS_031_B2B_SIPv4_TLS_ MakeCall - ReceiveCall - EndCall with RTP - 33s . . 4-40
VS_032_DUT_SIPv4_TLS_MakeCall - ReceiveCall with Registration. . . . . .4-40
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VS_033_B2B_SIPv4_TLS_ MakeCall - ReceiveCall - EndCall with SRTP - 33s .4-41
SIPv4_UDP_Basic_Call_without_RTP_Max_CPS . . . . . . . . . . . . . . . . . . . . 4-41
SIPv4_TCP_Basic_Call_without_RTP_Max_CPS . . . . . . . . . . . . . . . . . . . . 4-41
SIPv4_TLS_Basic_Call_without_RTP_Max_CPS. . . . . . . . . . . . . . . . . . . . . 4-41
SIPv4_UDP_Basic_Call_with_RTP_1sec_Max_CPS . . . . . . . . . . . . . . . . . . 4-42
SIPv4_TCP_Basic_Call_with_RTP_1sec_Max_CPS . . . . . . . . . . . . . . . . . . 4-42
SIPv4_TLS_Basic_Call_with_RTP_1sec_Max_CPS . . . . . . . . . . . . . . . . . . 4-42
SIPv4_UDP_Basic_Call_with_RTP_30sec_Max_CPS . . . . . . . . . . . . . . . . . 4-42
SIPv4_TCP_Basic_Call_with_RTP_30sec_Max_CPS . . . . . . . . . . . . . . . . . 4-42
SIPv4_TLS_Basic_Call_with_RTP_30sec_Max_CPS . . . . . . . . . . . . . . . . . 4-42
SIPv4_UDP_Basic_Call_with_RTP_3min_Max_CPS. . . . . . . . . . . . . . . . . . 4-43
SIPv4_TCP_Basic_Call_with_RTP_3min_Max_CPS . . . . . . . . . . . . . . . . . . 4-43
SIPv4_TLS_Basic_Call_with_RTP_3min_Max_CPS . . . . . . . . . . . . . . . . . . 4-43
SIPv4_UDP_Basic_Call_with_RTP_Max_Sessions . . . . . . . . . . . . . . . . . . . 4-43
SIPv4_TCP_Basic_Call_with_RTP_Max_Sessions . . . . . . . . . . . . . . . . . . . 4-43
SIPv4_TLS_Basic_Call_with_RTP_Max_Sessions . . . . . . . . . . . . . . . . . . . 4-44
SIPv4_UDP_Basic_Call_with_SRTP_Max_Sessions . . . . . . . . . . . . . . . . . . 4-44
SIPv4_TCP_Basic_Call_with_SRTP_Max_Sessions . . . . . . . . . . . . . . . . . . 4-44
SIPv4_TLS_Basic_Call_with_SRTP_Max_Sessions . . . . . . . . . . . . . . . . . . 4-44
SIPv4_Proxy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-44
SIPv4_B2BUA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-46
VoIPSIP with MSRP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-46MSRP_01_SIPv4_UDP_Text_Bidirectional . . . . . . . . . . . . . . . . . . . . . . . . . 4-46
MSRP_02_SIPv4_UDP_File_Transfer_Uni_aSDP . . . . . . . . . . . . . . . . . . . . 4-48
MSRP_03_SIPv4_UDP_File_Transfer_Uni_customSDP . . . . . . . . . . . . . . . 4-49
MSRP_04_SIPv4_UDP_Simultaneous_File_Text_ Transfer . . . . . . . . . . . . 4-50
MSRP_05_SIPv4_UDP_Simultaneous_Voice_Text_Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-50
MSRP_06_SIPv4_UDP_Simultaneous_Voice_File_Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-53
VoIPSIP Cloud Test Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-53
SC_001_B2B_SIPv4_T1_MakeCall_from_Cloud . . . . . . . . . . . . . . . . . . . . . 4-54
SC_002_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_n_to_n . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-55
SC_003_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_1_to_n . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-57
SC_004_B2B_SIPv4_T1_MakeCall_from_Cloud_w_Registration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-57
SC_005_B2B_SIPv4_T1_ReceiveCall_by_Cloud. . . . . . . . . . . . . . . . . . . . . 4-61
SC_006_B2B_SIPv4_T1_ReceiveCall_by_Cloud_RTP_n_to_n . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-62
SC_007_B2B_SIPv4_T1_Receive_Call_by_Cloud_RTP_n_to_1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-64
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w_Reg_to_DUT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-65
SC_009_DUT_SIPv4_T1_MakeCall_from_Cloud_w_Reg_to_DUT_RTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-66
SC_011_B2B_SIPv4_T2_BasicCall_between_two_Clouds . . . . . . . . . . . . . .4-69
SC_012_B2B_SIPv4_T2_BasicCall_between_two_Clouds_no_Routes . . . .4-71
SC_013_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n. . .4-71
SC_014_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_1_to_1. . .4-73
SC_015_DUT_SIPv4_T2_BasicCall_between_two_Clouds . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-73
SC_016_DUT_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-74
SC_017_DUT_SIPv4_T2_BasicCall_between_two_Clouds_RTP_1_to_1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-76
SC_018_B2B_SIPv4_T2_MakeCall_ReceiveCall_IM_w_RTP_two_Clouds . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-76
SC_019_B2B_SIPv6_T1_MakeCall_from_Cloud . . . . . . . . . . . . . . . . . . . . . .4-79
SC_020_B2B_SIPv6_T1_MakeCall_from_Cloud_RTP_1_to_n. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-80
SC_021_B2B_SIPv6_T1_ReceiveCall_by_Cloud . . . . . . . . . . . . . . . . . . . . .4-80
SC_022_B2B_SIPv6_T2_BasicCall_between_two_Clouds . . . . . . . . . . . . . .4-80
SC_023_B2B_SIPv6_T2_BasicCall_between_two_Clouds_no_Routes . . . .4-80
SC_024_B2B_SIPv6_T2_BasicCall_between_two_Clouds_RTP_1_to_1. . .4-80
SC_025_B2B_SIPv4_T1_MakeCall_from_Cloud_SRTP_1_to_n . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-80
SC_026_B2B_SIPv4_T2_BasicCall_between_two_Clouds_SRTP_1_to_1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-80
VS_SMS_001_B2B_SIPv4 UE vs. CSCF Complete SMS Flow . . . . . . . . . . .4-81
VS_SMS_002_DUT_SIPv4 UE vs. UE End-to-end SMS Flow. . . . . . . . . . . .4-82
Skinny Procedures, Sample Test Configurations and Test Scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-84
Sample Test Configurations Naming Convention . . . . . . . . . . . . . 4-84Skinny Predefined Procedures . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-85Skinny Test Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-87
Skinny Signaling Only . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-87
SK_001_7902_SO_US_100_Chs_IPv4_Static_Seq_Registration_5_retries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-87
SK_002_7960_SO_US_100_Chs_IPv4_Static_Seq_Registration_5_retries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-88
SK_005_7902_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-88
SK_006_7960_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-89
SK_003_7902_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop_5_retries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-90
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SK_004_7960_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop_5_retries. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-91
SK_007_7902_SO_US_100_Chs_IPv4_Static_Seq_Reg_5s_Sleep_Dereg_5_retries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-91
SK_008_7960_SO_US_100_Chs_IPv4_Static_Seq_Reg_5s_sleep_Dereg_5_retries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-92
Skinny Bulk Calls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-93
SK_009_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_10s . . . . . 4-93
SK_010_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_3min . . . . 4-95
SK_011_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_30min . . . 4-95
SK_012_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_10s . 4-95
SK_013_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-97
SK_014_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_30min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-97
SK_016_7902_SM_US_300_Chs_IPv4_Static_Basic_Call_DTMFs_inband_3min_7902 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-97
SK_017_7902_SM_US_300_Chs_IPv4_Static_Basic_Call_DTMFs_out-of-band_3min_7902. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-99
SK_021_7902_SM_US_300_Chs_IPv4_Static_BasicCall_Tone_highFreq_inband_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-99
SK_020_7902_SM_US_300_Chs_IPv4_Static_BasicCall_Tone_medFreq_inband_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-101
SK_019_7902_SM_US_300_Chs_IPv4_Static_BasicCall_Tone_lowFreq_inband_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-101
SK_015_7902_SM_US_10K_BHCA_IPv4_Static_Basic_Call_Voice_1min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-101
SK_018_7902_SM_US_25K_BHCA_IPv4_Static_Basic_Call_DTMFs_inband_3_min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-103
Advanced Call Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-105
SK_022_7960_SM_US_5_Chs_IPv4_Static_Hold_Resume . . . . . . . . 4-105
SK_031_7960_SM_US_5_Chs_IPv4_Static_List_Ad_Hoc_Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-106
SK_032_7960_SM_US_5_Chs_IPv4_Static_Forward_All_Calls . . . . . 4-108
SK_033_7960_SM_US_5_Chs_IPv4_Static_Forward_Busy . . . . . . . . 4-109
SK_034_7960_SM_US_5_Chs_IPv4_Static_Forward_No_Answer. . . 4-111
SK_026_7960_SM_US_5_Chs_IPv4_Static_Ad_hoc_Conference . . . 4-113
SK_027_7960_SM_US_5_Chs_IPv4_Static_MeetMe_Conference . . . 4-114
SK_028_7960_SM_US_5_Chs_IPv4_Static_Join_2_Calls . . . . . . . . . 4-115
SK_024_7960_SM_US_5_Chs_IPv4_Static_Blind_Transfer . . . . . . . . 4-117
SK_023_7960_SM_US_5_Chs_IPv4_Static_Transfer_with_Consultation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-119
SK_025_7960_SM_US_5_Chs_IPv4_Static_Direct_Transfer_of_2_parties_on_a_line. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-120
SK_036_7960_SM_US_5_Chs_IPv4_Static_Call_Group_Pickup . . . . 4-121
SK_037_7960_SM_US_5_Chs_IPv4_Static_Call_Pickup . . . . . . . . . . 4-123
Mixed Skinny and SIP - SIP UAs. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-125
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MIX_023_7960_S0_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_10s 4-125
MIX_024_7960_SO_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-128
MIX_025_7960_SO_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_30_min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-128
MIX_026_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_SK_Call_10s . . . . 4-128
MIX_027_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_SK_Call_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-131
MIX_028_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_SK_Call_30min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-131
MIX_020_7960_SM_US_900_Chs_IPv4_Static_SIP_to_SK_Call_Voice_10s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-131
MIX_021_7960_SM_US_900_Chs_IPv4_Static_SIP_to_SK_Call_Voice_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-134
MIX_022_7960_SM_US_900_Chs_IPv4_Static_SIP_to_SK_Bulk_Call_Voice_30_min. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-134
MIX_017_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_Call_Voice_10s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-135
MIX_018_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_Call_Voice_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-138
MIX_019_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_Call_Voice_30_min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-138
MIX_031_7960_SO_US_75k_BHCA_IPv4_Static_Sk_to_SIP_Call . . .4-139
MIX_029_7960_SM_US_75k_BHCA_IPv4_Static_Sk_to_SIP_Call_Voice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-142
MIX_032_7960_SO_US_75k_BHCA_IPv4_Static_SIP_to_Sk_Call . . .4-142
MIX_030_7960_SM_US_75k_BHCA_IPv4_Static_SIP_to_Sk_Call_Voice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-145
Mixed Skinny and SIP - SIP Trunk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-146
MIX_001_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_10s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-146
MIX_002_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-149
MIX_003_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_30min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-149
MIX_004_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_10s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-149
MIX_005_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-152
MIX_006_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_30min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-152
MIX_012_7960_SM_US_900_Chs_IPv4_Static_SIP_to_SK_trunk_Call_Voice_10s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-152
MIX_013_7960_SM_US_900_Chs_IPv4_Static_SIP_SK_trunk_Call_Voice_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-155
MIX_014_7960_SM_US_900_Chs_IPv4_Static_SIP_SK_trunk_Call_Voice_30min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-155
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MIX_009_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunkCall_Voice_10s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-155
MIX_010_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunk_Call_Voice_3min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-158
MIX_011_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunk_Call_Voice_30min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-158
MIX_008_7960_SO_US_80k_BHCA_IPv4_Static_SIP_to_SK_trunk_Bulk_Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-158
MIX_016_7960_SM_US_75k_BHCA_IPv4_Static_SIP_to_SK_trunk_Call_Voice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-161
MIX_007_7960_SO_US_80k_BHCA_IPv4_Static_SK_to_SIP_trunk_Bulk_Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-164
MIX_015_7960_SM_US_75k_BHCA_IPv4_Static_Sk_to_SIP_trunk_Call_Voice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-167
H.323 Sample Test Configurations and Test Scenarios. . 4-169VoIP H.323 Test Configurations . . . . . . . . . . . . . . . . . . . . . . . . . .4-169
VH_001_B2B_H323v4_NC_Basic_Call . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-170
VH_002_B2B_H323v4_NC_Basic_Call_with_RTP. . . . . . . . . . . . . . . . . . . 4-171
VH_003_B2B_H323v4_FC_Basic_Call_with_RTP . . . . . . . . . . . . . . . . . . . 4-172
VH_004_B2B_H323v4_PC_Basic_Call_with_RTP. . . . . . . . . . . . . . . . . . . 4-172
VH_005_B2B_H323v4_T_Basic_Call_with_RTP . . . . . . . . . . . . . . . . . . . . 4-173
VH_006_B2B_H323v4_NC_Make_Call_with_RTP. . . . . . . . . . . . . . . . . . . 4-173
VH_007_B2B_H323v4_FC_Make_Call_with_RTP . . . . . . . . . . . . . . . . . . . 4-173
VH_008_B2B_H323v4_PC_Make_Call_with_RTP. . . . . . . . . . . . . . . . . . . 4-174
VH_009_B2B_H323v4_NC_Receive_Call_with_RTP . . . . . . . . . . . . . . . . 4-174
VH_010_B2B_H323v4_FC_Receive_Call_with_RTP. . . . . . . . . . . . . . . . . 4-174
VH_011_B2B_H323v4_PC_Receive_Call_with_RTP. . . . . . . . . . . . . . . . . 4-175
VH_012_B2B_H323v4_NC_Basic_Call_with_RTP_Trunk . . . . . . . . . . . . . 4-175
VH_013_B2B_H323v4_FC_Basic_Call_with_RTP_Trunk . . . . . . . . . . . . . 4-175
VH_014_B2B_H323v4_NC_Basic_Call_with_RTP_all_codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-175
VH_015_B2B_H323v4_NC_Basic_Call_with_RTP_QoV . . . . . . . . . . . . . . 4-175
VH_016_B2B_H323v4_NC_Basic_Call_with_HwRTP . . . . . . . . . . . . . . . . 4-175
VH_017_B2B_H323v4_NC_Basic_Call_with_HwRTP_10GA_Trunk . . . . . 4-176
VH_018_GK_H323v4_NC_Basic_Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-176
VH_019_GK_H323v4_NC_Basic_Call_with_RTP . . . . . . . . . . . . . . . . . . . 4-177
VH_020_GK_H323v4_FC_Basic_Call_with_RTP. . . . . . . . . . . . . . . . . . . . 4-178
VH_021_GK_H323v4_PC_Basic_Call_with_RTP . . . . . . . . . . . . . . . . . . . 4-178
VH_022_GK_H323v4_T_Basic_Call_with_RTP . . . . . . . . . . . . . . . . . . . . . 4-178
VH_023_GK_H323v4_NC_Basic_Call_with_RTP_Trunk . . . . . . . . . . . . . . 4-178
VH_024_GK_H323v4_FC_Basic_Call_with_RTP_Trunk . . . . . . . . . . . . . . 4-179
VH_025_GK_H323v4_NC_Basic_Call_with_RTP_all_codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-179
VH_026_GK_H323v4_NC_Basic_Call_with_RTP_QoV . . . . . . . . . . . . . . . 4-179
VH_027_GK_H323v4_NC_Basic_Call_with_HwRTP . . . . . . . . . . . . . . . . . 4-180
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VH_028_GK_H323v4_NC_Basic_Call_with_HwRTP_10GA_ER_Trunk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-180
H.248 Sample Test Configurations and Test Scenarios . . 4-181Used Test Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-181VoIP H.248 Test Configurations. . . . . . . . . . . . . . . . . . . . . . . . . . 4-182
VM_001_H248_IPv4_B2B_with_version_v3 . . . . . . . . . . . . . . . . . . . . . . . .4-182
VM_002_H248_IPv4_B2B_with_version_v2 . . . . . . . . . . . . . . . . . . . . . . . .4-184
VM_003_H248_IPv4_B2B_with_version_v1 . . . . . . . . . . . . . . . . . . . . . . . .4-184
VM_004_H248_IPv6_B2B_with_version_v3 . . . . . . . . . . . . . . . . . . . . . . . .4-184
VM_005_H248_IPv6_B2B_with version_v2 . . . . . . . . . . . . . . . . . . . . . . . . .4-184
VM_006_H248_IPv6_B2B_with_version_v1 . . . . . . . . . . . . . . . . . . . . . . . .4-184
VM_007_H248_IPv4_B2B_with_message_maximum_size_4000 . . . . . . . .4-184
VM_008_H248_IPv4_B2B_reply_send_AuditValue . . . . . . . . . . . . . . . . . . .4-184
VM_009_H248_IPv4_B2B_reply_send_AuditCapabilities . . . . . . . . . . . . . .4-186
VM_010_H248_IPv4_B2B_enable_retransmissions . . . . . . . . . . . . . . . . . .4-187
VM_011_H248_IPv4_B2B_Access_gw . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-187
VM_012_H248_IPv4_B2B_Border_gw. . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-189
VM_013_H248_IPv4_B2B_Trunking_gw . . . . . . . . . . . . . . . . . . . . . . . . . . .4-192
VM_014_H248_IPv4_B2B_Residential_gw . . . . . . . . . . . . . . . . . . . . . . . . .4-193
VM_015_H248_IPv4_B2B_MID_IP_Address . . . . . . . . . . . . . . . . . . . . . . . .4-195
VM_016_H248_IPv4_B2B_MID_Device_Name. . . . . . . . . . . . . . . . . . . . . .4-197
VM_017_H248_IPv4_B2B_MID_IPAddress_Port . . . . . . . . . . . . . . . . . . . .4-197
VM_018_H248_IPv4_B2B_MID_MGW_MGC_DNS_Name. . . . . . . . . . . . .4-197
VM_019_H248_IPv4_B2B_performance_topology2_rtp . . . . . . . . . . . . . . .4-197
VM_020_H248_IPv4_B2B_performance_topology1_rtp . . . . . . . . . . . . . . .4-200
VM_021_H248_IPv4_B2B_without_registration . . . . . . . . . . . . . . . . . . . . . .4-202
VM_022_H248_IPv4_B2B_G711_ulaw . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-202
VM_023_H248_IPv4_B2B_G711_alaw . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-204
VM_024_H248_IPv4_B2B_G723_1_5.3 . . . . . . . . . . . . . . . . . . . . . . . . . . .4-204
VM_025_H248_IPv4_B2B_G723_1_6.3 . . . . . . . . . . . . . . . . . . . . . . . . . . .4-204
VM_026_H248_IPv4_B2B_G726_16 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-205
VM_027_H248_IPv4_B2B_G726_24 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-205
VM_028_H248_IPv4_B2B_G726_32 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-205
VM_029_H248_IPv4_B2B_G726_40 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-205
VM_030_H248_IPv4_B2B_G729AB. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-205
VM_031_H248_IPv4_B2B_ILBC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-205
VM_032_H248_IPv4_B2B_auto_sdp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-205
VM_033_H248_IPv4_B2B_custom_SDP . . . . . . . . . . . . . . . . . . . . . . . . . . .4-207
VM_034_H248_IPv4_B2B_sdp_renegociation. . . . . . . . . . . . . . . . . . . . . . .4-208
VM_035_H248_IPv4_vs_DUT_RGW_analog_basic_call. . . . . . . . . . . . . . .4-208
VM_036_H248_IPv4_vs_DUT_RGW_analog_basic_all_with_renegotiation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-208
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MGCP Sample Test Configurations and Test Scenarios . 4-210Used Test Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-210VoIP MGCP Test Configurations. . . . . . . . . . . . . . . . . . . . . . . . . .4-211
VMG_001_MGCP_IPV4_B2B_Basic_Call_Gw_vs_CA_with_RTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-211
VMG_002_MGCP_IPV4_B2B_Basic_Call_Gw_vs_CA_signalling_only . . 4-212
VMG_003_MGCP_IPV4_B2B_Basic_Call_GW1_calls_GW2_through_CA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-213
VMG_004_MGCP_IPV4_B2B_Basic_Call_with_Renegociation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-215
VMG_005_MGCP_IPV4_B2B_Basic_Call_with_DTMFs . . . . . . . . . . . . . . 4-216
VMG_006_MGCP_IPV4_B2B_Basic_Call_with_voice_session_G711_Alaw. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-218
VMG_007_MGCP_IPV4_B2B_Basic_Call_with_voice_session_G726-40 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-220
VMG_008_MGCP_IPV4_B2B_Basic_Call_with_RSIP_from_scenario. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-220
VMG_009_MGCP_IPV4_B2B_Basic_Call_with_strip_leading_zero_enabled . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-220
VMG_010_MGCP_IPV4_VS_DUT_BTS_Basic_Call_with_BTS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-222
VMG_011_MGCP_IPV4_B2B_HWRTP_Non_Agg_Basic_Call_RTP_8000ch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-223
VMG_012_MGCP_IPV4_B2B_HWRTP_1G_Agg_Basic_Call_RTP_8000ch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-225
VMG_013_MGCP_IPV4_B2B_HWRTP_10G_Agg_Basic_Call_RTP_96000ch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-225
PSTN Sample Test Configurations and Test Scenarios. . 4-228Used Test Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-228PSTN Test Configurations. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4-228
PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS . . . . . . . . . . . . . . . . . . . . . . . . 4-229
PSTN_002_B2B_T1_CAS_FGD_D4_B8ZS . . . . . . . . . . . . . . . . . . . . . . . . 4-229
PSTN_003_B2B_E1_CAS_Argentina_G704_HDB3. . . . . . . . . . . . . . . . . . 4-230
PSTN_004_B2B_T1_4ESS_D4_B8ZS . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-230
PSTN_005_B2B_T1_QSIG_D4_B8ZS . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-230
PSTN_006_B2B_T1_5ESS_D4_B8ZS . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-230
PSTN_007_B2B_E1_ISDN_QSIG_G704_HDB3 . . . . . . . . . . . . . . . . . . . . 4-230
PSTN_008_B2B_E1_ISDN_KHT_G704_HDB3 . . . . . . . . . . . . . . . . . . . . . 4-231
PSTN_009_VS_Cisco_E1_ISDN_vs_E1_ISDN . . . . . . . . . . . . . . . . . . . . . 4-231
PSTN_010_VS_Cisco_SIP_vs_T1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-232
PSTN_011_VS_Cisco_T1_CAS_IMM_vs_T1_CAS_IMM . . . . . . . . . . . . . 4-233
PSTN_012_VS_Cisco_T1_ISDN_5ESS_vs_T1_ISDN_5ESS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-233
PSTN_013_VS_Cisco_T1_vs_SIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-233
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Chapter 5 Extended Functionality SIP Tests Suite
SIP Tests Suite Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-1General Test Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-1
General Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-2
Network Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-2
Activity Settings. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-2
Scenario Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-3
Test Objective Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-3
Test Categories . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-4Customizable Test Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-4
Predefined Common Test Procedures . . . . . . . . . . . . . . . . . 5-5Register . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-5Make Call. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-6UnRegister. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-7
Test Descriptions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-8Registration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-8
Register. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-8
Basic Calls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-8Basic_Call_Complete_Audio_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-8
Basic_Call_Complete_Audio_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-9
Basic_Call_Complete_Multimedia_AC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-9
Basic_Call_Complete_Multimedia_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-9
Basic_Call_Busy_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-9
Basic_Call_Busy_MCH. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-9
Basic_Call_Complete_Cancel_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-10
Basic_Call_Complete_Cancel_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-10
Advanced SIP Features. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-10Call_Hold_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-10
Call_Hold_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-11
Consultation_Hold_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-12
Consultation_Hold_MCH. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-13
Music_on_Hold_LPS. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-13
Music_on_Hold_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-14
Transfer_Unattended_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-14
Transfer_Unattended_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-15
Transfer_Attended_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-16
Transfer_Attended_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-17
Transfer_Instant_Messaging_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-17
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Transfer_Instant_Messaging_MCH. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-18
Call_Forwarding_Unconditional_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-19
Call_Forwarding_Unconditional_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-19
Call_Forwarding_on_Busy_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-20
Call_Forwarding_on_Busy_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-20
Call_Forwarding_on_No_Answer_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-21
Call_Forwarding_on_No_Answer_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-21
Three_Way_Conference-Third_Party_Added_LPS . . . . . . . . . . . . . . . . . . . 5-21
Three_Way_Conference-Third_Party_Added_MCH . . . . . . . . . . . . . . . . . . . 5-22
Three_Way_Conference-Third_Party_Joins_LPS . . . . . . . . . . . . . . . . . . . . 5-23
Three_Way_Conference-Third_Party_Joins_MCH . . . . . . . . . . . . . . . . . . . . 5-23
Single_Line_Extension_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-24
Single_Line_Extension_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-25
Find-Me_LPS. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-25
Find-Me_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-27
Call_Park_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-27
Call_Park_MCH. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-28
Call_Pickup_LPS. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-28
Call_Pickup_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-29
Automatic_Redial_LPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-29
Automatic_Redial_MCH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-30
Click_to_Dial_LPS. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-30
Appendix A Creating a SIP Message from Template
SIP Message Elements . . . . . . . . . . . . . . . . . . . . . . . . . . . . A-2Specific Request Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A-2Specific Response Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . A-3Common Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A-4
Appendix B The Expression Evaluator Syntax
Data Types. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-1
Operators . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-2Arithmetic Operators . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-2Relational Operators . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-3Logical Operators . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-4Format Operator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-5Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-7
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Compound Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-8
Appendix C Using the H248 Descriptor Editor
The Descriptor Editor GUI . . . . . . . . . . . . . . . . . . . . . . . . . . .C-1Editing Transmitted Request Messages . . . . . . . . . . . . . . . . . . . . . C-2
Non-Leaf Items . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C-2
Leaf Items . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C-3
Editing Expected Response Messages . . . . . . . . . . . . . . . . . . . . . . C-6
Appendix D Using the MGCP Parameter Editor
MGCP Script Functions Overview. . . . . . . . . . . . . . . . . . . . .D-1Send Type Functions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D-2
Tx Command Page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D-2
Rx Response Page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D-3
Wait Type Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D-4Rx Command Page. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D-4
Tx Response Page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D-4
Appendix E Skinny Sample Configurations Overview
Appendix E Support for Multipart SIP Messages
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1Chapter 1: Introduction to the VoiceTest Libraries
This chapter covers the following topics:
• Who Should Read This Manual on page 1-1.
• How This Manual Is Organized on page 1-1.
• Voice Test Libraries Overview on page 1-2.
• Voice Libraries Function Sets on page 1-2.
Who Should Read This Manual
This manual is intended for users intending to edit VoIP and PSTN tests using the Voice Test Libraries of the IxLoad Voice Plug-In application.
This manual describes the Voice Test Libraries, their sets of script functions, their associated global and local parameters. The manual also includes a descrip-tion of the predefined Voice scenario variables that can be used within Voice test scenarios.
How This Manual Is Organized
This manual is organized as follows:
• Chapter 1, Introduction to the Voice Test Libraries
• Chapter 2, Voice Test Libraries Settings
• Chapter 3, Voice Functions Reference
• Chapter 4, Basic Test Scenarios and Procedures
• Chapter 5, Extended Functionality SIP Tests Suite
• Appendix A, Creating a SIP Message from Template
• Appendix B, The Expression Evaluator Syntax
• Appendix C, Using the H248 Descriptor Editor
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• Appendix D, Using the MGCP Parameter Editor
• Appendix E, Skinny Sample Configurations Overview
Voice Test Libraries Overview
The VoIP SIP Test Library includes test functions for establishing calls using the SIP signaling protocol, and for generating RTP media streaming across estab-lished calls.
The VoIP Test Skinny Library functions assemble fully featured SCCP clients by simulating Cisco SKINNY phones that generate and receive Skinny messages, and generate RTP media streaming across calls.
In addition to SIP, Skinny, and RTP functions, test flows use special call control functions from the Flow Test Library that indicate the start/end of an execution flow, test for variable status at a particular moment in the flow, and so on. Flow functions are fully compatible with the functions in the SIP, Skinny, and RTP libraries and can be used conjointly with these functions in building test scenar-ios.
The functions in the Voice test libraries are grouped by protocols as follows:
• VoIP SIP Functions Library on page 3-1.
• VoIP Skinny Functions Library on page 3-41
• VoIP Media Functions Library on page 3-86.
• VoIP Flow Functions Library on page 3-111.
• VoIP H323 RAS Library on page 3-116
• VoIP H323 Functions Library on page 3-118
• VoIP H248 Functions Library on page 3-123
• VoIP MGCP Functions Library on page 3-138
• Digital T1/E1 Functions Library on page 3-155
Voice Libraries Function Sets
Table 1-1 to Table 1-6 describe the script functions currently implemented in the Voice test libraries.
SIP Library Functions Set
Table 1-1 briefly describes the functions implemented in the VoIP SIP test library.
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Skinny Library Functions Set
Table 1-1 briefly describes the functions implemented in the VoIP Skinny test library.
Table 1-1. SIP Library - Functions Description
Function Description
Send Request Sends a SIP request.
Send Response Sends a SIP response.
Wait Request Waits for a SIP request.
Wait Response Waits for a SIP response.
Wait Message Waits for a SIP message and matches it with one of the message templates defined within the function.
Retransmit Last Message
Re-sends the last transmitted SIP message.
Extract Variables Extracts scenario variables from SIP messages.
Table 1-2. Skinny Library - Functions Description
Function Description
OffHook Notifies the Cisco CallManager that a terminal is in an off-hook condition,
OnHook Notifies the Cisco CallManager that a station is now in an on-hook condition.
NewCall Sends a SoftKeyEvent message to the Cisco CallManager, requesting dial tone.
EndCall Sends a SoftKeyEvent message to the Cisco CallManager, requesting a specific call completion.
MakeCall Originates a call by dialing the phone number and performing the call establishment.
WaitCall Waits for an incoming call.
AnswerCall Answers an incoming call by going off-hook and performing the call establishment.
DialDigits Dials the specified digits.
WaitDigits Waits for a specified digits sequence.
HoldCall Performs a hold operation on a specified call reference.
RetrieveCall Performs a retrieve operation on a specified call reference.
SetupXfer Initiates a transfer or a conference procedure.
CompleteXfer Completes a transfer or a conference procedure.
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RTP Library Functions Set
Table 1-3 briefly describes the functions implemented in the VoIP RTP test library.
Transfer As a combination of the SetupXfer and CompleteXfer functions, it sets up a transfer or a conference.
ForwardAllCalls Sends a SoftKeyEvent message to the Cisco CallManager, requesting that all incoming calls be forwarded.
ParkCall Parks a specific call.
RegisterClient Registers the Skinny client with a Cisco CallManager.
UnregisterClient Unregisters the Skinny client from a Cisco CallManager.
GetCallInfo Retrieves the call information into the predefined VoIP Skinny variables.
MeetMe Sets up a MeetMe type conference call.
RemoveLastConferenceParty
Removes from a conference call the party that has joined last.
SendStimulus A Skinny Client uses this message to inform the Cisco CallManager that a functional stimulus button was clicked.
SendSoftKey Emulated stations of the CP-7940/60 type use this message to inform the Cisco CallManager of a softkey event.
IsSoftKeyAvailable Verifies if a certain soft key is available.
WaitForEvent Is used by a Skinny Client to search for a specified message in the message queue. If the message does not exist, the function waits for the specified message.
Table 1-2. Skinny Library - Functions Description
Function Description
Table 1-3. RTP Library - Functions Description
Function Description
Generate DTMF Generates a specified DTMF sequence.
Detect DTMF Detects a sequence of DTMF signals.
Generate MF Generates a specified MF sequence.
Detect MF Detects a sequence of MF signals.
Generate Tone Generates a custom tone.
Wait For Tone Detects a custom tone.
RTP Control Checks for RTP completion or terminates a RTP script function.
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Flow Functions Set Table 1-4 listst the functions implemented in the Voice Flow test library.
Path Confirmation Executes a Path Confirmation sequence using DTMFs, MFs, or Custom Tones.
Talk Plays the specified wave files across the established call.
Listen Listens to the specified wave files across the established call.
Voice Session Plays and records simultaneously the specified wave files across the established call.
Multimedia Session Plays an audio file and a video MP4 file across en established SIP or H.323 call.
T38 Fax Session Sends or receives a specified image file across a fax session negotiated using the SIP signaling protocol.
Table 1-3. RTP Library - Functions Description
Function Description
Table 1-4. Flow and Analysis Library - Functions Description
Function Description
Start Indicates the beginning of a Test Scenario flow on the channel.
Stop Indicates the end of an execution thread or the end of the entire Test Scenario.
Variable Set Declares and sets the variable values to use in the test scenario.
Variable Test Assesses a series of logical variables.
Sleep Applies a static or random delay to the execution flow.
Procedure Declares a procedure in the current test scenario.
Exit Proc Determines the output of a procedure.
Counter Op Inserts user-defined statistic counters, which may be incremented, decremented, or reset.
Test Time Assesses the current time.
Log Message Enables you to edit messages to include in the execution log.
Dump Variable Generates a log containing all variables from all engines and the user-defined variables.
Error Handler This script function can be used to minimize the number of connectors in a script.
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MGCP Functions Set
Table 1-5 listst the functions implemented in the VoIP MGCP test library.
Table 1-5. MGCP Library - Functions Description
Function Description
Send NTFY The Send Notify script function implements the transaction initiated by a Notify command.
Send DLCX (GW) The Send DLCX script function implements the transaction initiated by a sent DeleteConnection command.
Send RSIP The Send RSIP script function implements the transaction initiated by a RestartInProgress (RSIP) command sent from the MGW to the CA.
Wait CRCX The Wait CRCX script function implements the transaction initiated by the receiving of a CreateConnection (CRCX) command, followed by the sending of a response to this command.
Wait DLCX The Wait DLCX script function implements the transaction initiated by the receiving of a DeleteConnection (CRCX) command, followed by the sending of a response to this command.
Wait MDCX The Wait MDCX script function implements the transaction initiated by a the receiving of a ModifyConnection (CRCX) command, followed by the sending of a response to this command.
Wait RQNT The Wait RQNT script function implements the transaction initiated by the receiving of a RequestNotify (RQNT) command, followed by the sending of a response to this command.
Wait AUEP The Wait AUEP script function implements the transaction initiated by the receiving of an AuditEndpoint (AUEP) command, followed by the sending of a response to this command.
Wait AUCX The Wait AUCX script function implements the transaction initiated by the receiving of an AuditConnection (AUCX) command, followed by the sending of a response to this command.
Wait EPCF The Wait EPCF script function implements the transaction initiated by the receiving of an EndpointConfiguration (EPCF) command, followed by the sending of a response to this command.
Wait Command (GW)
The Wait Command (GW) function specifies one or more MGCP commands awaited by the MGW. If any of these commands is received, the user-configured response to the command is sent and, if executed successfully, the function exits on the output identifying the matched command.
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T1/E1 Functions Set Table 1-6 listst the functions implemented in the Digital T1/E1 test library.
Send RQNT The Send RQNT script function implements the transaction initiated by a RequestNotify (RQNT) command sent by the CA
Send CRCX The Send CRCX script function implements the transaction initiated by a CreateConnection (CRCX) command sent by the CA.
Send DLCX The Send DLCX script function implements the transaction initiated by a DeleteConnection (DLCX) command sent by the CA.
Send MDCX The Send MDCX script function implements the transaction initiated by a Mo-difyConnection (MDCX) command sent by the CA.
Send AUCX The Send AUCX script function implements the transaction initiated by an AuditConnection (AUCX) command sent by the CA.
Send AUEP The Send AUEP script function implements the transaction initiated by a Audit Connection (AUEP) command sent by the CA.
Send EPCF The Send EPCF script function implements the transaction initiated by a Endpoint Configuration (EPCF) command sent by the CA.
Wait NTFY The Wait NTFY script function implements the transaction initiated by the receiving of an MGCP Notify command sent by an MGW.
Wait DLCX The Wait DLCX script function implements the transaction initiated by the receiving at the CA of an MGCP DeleteConnection (DCLX) command sent by an MGW.
Wait Command (CA)
The Wait Command (CA) script function specifies one or more MGCP commands awaited by the CA. If any of these commands is received, the user-configured response to the command is sent and, if executed successfully, the function exits on the output identifying the matched command.
Wait RSIP The Wait RSIP script function implements the transaction initiated by the receiving at the CA of an MGCP RestartInProgress (RSIP) command sent by an MGW.
Table 1-5. MGCP Library - Functions Description (Continued)
Function Description
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Table 1-6. Digital T1/E Library - Functions Description
Function Description
Make Call This script function initiates a call to the specified destination.
Receive Call This script function answers an incoming call.
End Call This script function terminates an established call.
Path Confirmation This script function executes a Path Confirmation sequence, wherein the path confirmation initiator sends a specific digit (DTMF/MF/tone) sequence and then waits to receive another digit (DTMF/MF/Tone) sequence from the remote party.
Talk This script function plays back a wave file from the IxLoad Wave Files pool.
Listen This script function allows the recording of a wave file for a specified duration.
Voice Session This script function plays back a wave file and records a wave file at the same time.
Generate DTMF This script function generates a sequence of Dual Tone Multiple Frequency (DTMF) signals.
Detect DTMF This script function is used to detect a sequence of DTMF signals.
Generate MF This script function generates a sequence of Multi - Frequency (MF) tones.
Detect MF This script function is used to detect a sequence of MF tones.
Generate Tone This script function generates a single custom tone (single or dual continuous, or cadence) that can be selected from the Custom Tones Pool.
Wait for Tone This script function detects any custom tone from a user-configured tone list.
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2Chapter 2: Voice Test Libraries Settings
This chapter covers the following topics:
• Global Settings on page 2-1.
Global Settings
The Global Settings represent application-wide, default test execution parameters associated with the IxLoad Voice Plug-in environment and the test libraries sup-ported by it. Configured Global Settings are taken into account, for example, when placing a function in the Scenario Editor, when running a test execution session and when using the application workspace.
The Global Settings window is accessed by clicking the button in the Scenario Editor window. The following Global Settings categories represent Voice test library settings:
• Library Settings and Outputs: This category represents the global settings for the most frequently used parameters of the functions in the Voice library. A Voice script function that is initially placed in the Scenario Editor can be con-figured to use a subset of these library settings.
• Scenario Editor Defaults: This category specifies the default settings for each SIP, Skinny, H.323, MGCP, H.248/MEGACO, Digital T1/E1, RTP, and Flow script function that has been placed in the Scenario Editor.
Configuring Library Settings and Outputs
Choose Library Settings and Outputs in the left pane of the Global Settings window to configure a subset of the VoIP and T1/E1 test library parameters. This settings category also enables you to define the default output names for Voice library functions.
Note: These settings are used by a VoIP or T1/E1 script function placed in the Scenario Editor only if the function’s Use Global Settings parameter is selected.
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RTP Settings
Table 2-1 lists the global VoIP RTP library settings.
Table 2-1. VoIP RTP Library Global Settings
Name Description
RTP Transmission Mode
Specifies the RTP transmission mode, which can be one of the following:
• In Band (using RTP media streaming)
• Out of Band – Using 2833 EVENT Payload Format
• Out of Band – Using 2833 TONE Payload Format
RTP Playback
Play x time(s) Sets the number of times the wave file is played.
Repeat Continuous for
Sets the time period the wave file is played (in seconds, minutes, or hours).
Use Talk Time (only for BHCA/CPS objective)
In the case of a test configured with a BHCA or CPS objective, choosing this option plays back the wave file for the duration of the talk time parameter.
Audio Playback
Output Volume Sets the volume level during the VoIP communication. The available values include:
• -15 dB (default)
• -20 dB
• -25 dB
• -30 dB
Path Confirmation: Specifies the DTMF/MF/Tone generation and detection settings for the RTP Path Confirmation function.
DTMF/MF/TONE Generation
Tone Duration The time duration (in ms) of a single tone. The range of values is 40 to 59960 ms. The default value is 200 ms.
Inter Tone Interval
The maximum amount of time (in ms) between two consecutive generated DTMFs. The range of values is 40 to 59960 ms. The default value is 200 ms.
Note: The sum of the tone duration and the inter tone interval is required to be less than 60000 ms.
Tone Amplitude The attenuation (in dB) of the DTMF tone. The minimum attenuation is 0 dB (no attenuation) and the maximum is –40 dB. The default value is –10 dB.
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Skinny Settings
Table 2-3 lists the global Skinny library settings.
DTMF/MF/TONE Detection
First Sequence Timeout
The time (in ms) allowed for receiving the first digit. After this period elapses, the function exits on Timeout output. The range of values is 200 to 999999 ms. The default value is 5000 ms.
Inter sequence Timeout
The maximum amount of time (in ms) allowed between consecutive DTMFs/MFs/custom tones sequences for a proper detection. After this period elapses, the function exits on Timeout output. The range of values is 200 to 999999 ms. The default value is 5000 ms.
QoV - When performing QoV computations, adding silence periods and increasing the liste duration enables the application to correctly resolve loops and identify the played clip.
Talk - Add extra silence
If selected, this adds to the Talk function a silence period of the duration specified in the adjacent field.
Listen - Increase duration by
If selected, this adds to the Listen function a silence period of the duration specified in the adjacent field.
Path Confirmation Sequence
Execute once The path confirmation sequence is executed once.
Execute for The path confirmation sequence is executed for a specified time interval.
Table 2-1. VoIP RTP Library Global Settings (Continued)
Name Description
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Table 2-2. VoIP Skinny Library Global Settings
Name Description
SoftKeys
The editable table displays an available softkeys list.
The following operations can be performed on the softkeys table:
Adds a softkey table entry.
Deletes a softkey table entry.
Displays a softkey’s properties.
Imports softkeys from a user-specified file.
Exports softkeys to a user-specified file.
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SIP Settings
Table 2-3 lists the global SIP library settings.
Table 2-3. VoIP SIP Library Global Settings
Name Description
Transport Layer
Disable TCP Support If this checkbox is enabled, support for the TCP protocol is disabled. The default value is Disabled.
Note: You need to disable TCP support, for example, when performing SIP performance testing on UDP transport.
Authorization
Preferred QOP Specifies the quality of protection mode to use when the server offers multiple choices. Available choices are:
• auth(entication) (default)
• auth(entication)-int(egrity)
Treat parser warnings as errors
If selected, the SIP parser warnings are considered errors. The default is Disabled.
Messages Queue Settings
Remove previously received messages
If set to Yes, clears the message queue of previously received messages. The default is No.
Limit queue size to maximum
If selected, this option allows you to indicate the maximum number of messages per channel that the queue can sustain.
Log For each voip peer activity, if any of the logs are enabled, a file is created and is named “voip_peer_$activityName.log”.
SIP sent Messages Check this option to log all SIP sent Messages.
SIP received Messages
Check this option to log all SIP received Messages.
SIP matched Messages
Check this option to log all SIP matched Messages.
Scenario Flow Path Check this option to log all Scenario functions entry and exit.
Timeline Events Check this option to log timeline events for each user.
Call States Check this option to log the SIP Call States.
RTP/SDP informations
Check this option to log all RTP/SDP events.
Call/Registration times
Check this option to log all SIP times.
Errors Check this option to log all SIP errors.
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STUN Settings
Since protocols such as SIP/RTP/RTCP use UDP packets for the transfer of sound, video, and/or real-time signaling traffic over the Internet, SIP endpoints operating from behind NAT devices have to use a STUN server in order to allow inbound media traffic. A STUN server enables endpoints behind a NAT device to first discover the presence and the type of a NAT, and then learn the address bindings allocated by the NAT.
The value of the Send STUN packets at every field defines the keep-alive time for the sent STUN packets.
T.38 Settings
Table 2-4 lists the global T.38 test library settings.
PSTN Settings
Table 2-5 through Table 2-8 list the global T1/E1 test library settings.
Log for channels Check this option to specify the channels that are logged.
Include timestamps Check this option to include timestamps for the logs.
Table 2-4. VoIP T.38 Library Global Settings
Name Description
T.38 Log Settings - This group specifies settings related to the collection of logs for T.38 traffic.
Enabled logs Specifies the logs to be collected when executing a T.38 Fax Session script function:
• T30 signals
• T38 signals
• Received image: If selected, received images are col-lected and stored on the Ixia ports in the /var/log folder following a naming pattern that includes the activity name, the user name, and the current loop, such as for example in /var/log/VoIPSipPeer1_recv_img_user003_loop001.tif
Log type Specifies if logs are collected for a number of loops (Loops option), or for a specified duration of time (Duration, expressed in minutes).
Range of Channels Specifies a range or channels for which logs are collected.
Table 2-3. VoIP SIP Library Global Settings
Name Description
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Table 2-5. PSTN Library - Global Timeouts
Parameter Description
Delay before execution
Specifies a global function execution delay value as either:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
No answer timeout Specifies the period of time for which the Make Call script function waits for an answer, after which the function exits on the Timeout output.
Wait other party to disconnect
if selected, the End Call script function does not initiate call termination itself, but waits for the other party to disconnect.
Wait for call timeout Specifies a period of time a Receive Call function waits for an incoming call, after which it exits on the Timeout output.
Completion timeout Specifies timeout value for the initiation of a call to the actual call established status.
Table 2-6. PSTN Library - ISDN
Parameter Description
Layer 1 bearer channel
The User info layer 1 parameter specified by ITU-T Recommendation Q.931, which can take the following values:
• ISDN_UIL1_G711ULAW
• ISDN_UIL1_G711ALAW
• OPERATOR_SPECIFIC
Bearer channel capability
The Information Transfer Capability parameter specified by ITU-T Recommendation Q.931, which can take the following values:
• BEAR_CAP_SPEECH
• BEAR_CAP_UNREST_DIG
• BEAR_CAP_DOT1K_AUDIO
Destination number type
The Called party number information element specified by ITU-T Recommendation Q.931, which can take the following values:
• Unknown
• International
• National
• Network specific
• Subscriber
• Abbreviated
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Destination number plan
The called party Numbering plan identification parameter specified by ITU-T Recommendation Q.931, which can take the following values:
• Unknown
• ISDN (default)
• Telephony
• National
• Private
Originating number type
The Caller party number information element specified by ITU-T Recommendation Q.931, which can take the following values:
• Unknown
• International
• National
• Network specific
• Subscriber
• Abbreviated
Originating number plan
The caller party numbering plan identification parameter specified by ITU-T Recommendation Q.931, which can take the following values:
• Unknown
• ISDN (default)
• Telephony
• National
• Private
Reject call reason Specifies a global call reject reason for the Receive Call script function. When this function rejects an incoming call, the configured reason is provided to the calling party.
Initiate RESTART procedure while shutting down spans
If selected, a RESTART procedure is initiated when the ISDN protocol is stopped. This procedure implies sending RESTART messages to the remote end (a single RESTART for the entire trunk or one RESTART for each B channel, depending on the variant) and waiting for RESTART ACKNOWLEDGE messages.
Table 2-7. PSTN Library - CAS
Parameter Description
Dial digits
Digit duration Specifies the digit duration.
Inter digit delay Specifies the inter-digit delay value.
Signal power Specifies the CAS signal power value.
Table 2-6. PSTN Library - ISDN
Parameter Description
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ANI/DNIS
Incoming call : Digits format
The digits format for incoming calls, which can be either of the following:
• DNIS
• DNIS/ANI
• ANI/DNIS
Incoming call : Calling party # of digits (ANI)
The expected number of digits that identify the calling party.
Important: If the calling party number is configured to a higher value at activity level, that value is used.
Outgoing call : Digits format
• The digits format for outgoing calls, which can be either of the following:
• DNIS
• DNIS/ANI
• ANI/DNIS
Timeouts
Wait for wink timeout The timeout value for receiving a wink.
Time between offhook and dial
The maximum period of time between the offhook signal and the first dialed digit.
Table 2-8. PSTN Library - Voice
Parameter Description
Playback
Play x time(s) The number of times the wave is played.
Repeat Continuous for
The period of time the wave is played for (in seconds, minutes, or hours).
Use Talk Time (all objectives except Channels)
In the case of a CPS objective, choosing this option plays back the wave for the duration of the talk time parameter.
Coding
Data format The voice encoding, any of the following: ALaw, MuLaw, PCM.
Sampling rate The voice sampling rate, 8000 or 11025 Hz.
Bits/sample The number of bits per voice sample, 8 or 16 bit.
DTMF/MF/Tones Generation
Tone Duration The time duration (in ms) of a single tone. The range of values is 50 to 10000 ms. The default value is 200 ms.
Table 2-7. PSTN Library - CAS
Parameter Description
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Function Outputs Results
The Function Outputs Results window enables you to define the global result res-olution (the output name—output result association) used by Voice functions.
All Voice functions placed in the Scenario Editor have a special Outputs tab that enables you to set the result resolution for each output. When a function output is set to the GLOBAL value, it takes the value corresponding to the specified out-put in the Function Outputs Results window.
To configure the global settings for each output:
1. Choosing Library Settings and Outputs> Function Output Results from the left tree pane in the Global Settings window (Figure 2-1) displays in the right pane a list of Function Outputs Results, where each output is character-ized by an Output Name and an Output Result.
2. You can perform the following operations:
• Add a new output.
• Modify an existing output – you are not allowed to modify the name of a predefined output.
• Delete an output – this operation can be performed only on the user-defined outputs. The predefined outputs cannot be removed.
Inter Tone Interval The maximum amount of time (in ms) between two consecutive generated DTMFs. The range of values is 50 to 10000 ms. The default value is 200 ms.
Tone Amplitude The attenuation (in dB) of the DTMF tone. The minimum attenuation is -3 dB (no attenuation) and the maximum is –54 dB. The default value is –3 dB.
DTMF/MF/Tones Detection
First tone timeout The time (in ms) allowed for receiving the first tone. After this period elapses, the function exits on the Timeout output. The range of values is 50 to 10000 ms. The default value is 3000 ms.
Max delay between tones
The maximum delay (in ms) between subsequent tones. elapses, the function exits on the Timeout output. The range of values is 50 to 10000 ms. The default value is 2000 ms.
Table 2-8. PSTN Library - Voice
Parameter Description
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Figure 2-1. Function Outputs Results
To add a new output:
1. Click the Add button. The Output Properties dialog opens.
2. Type the desired Output Name in the appropriate field.
3. Choose the Output Result. You have the following options:
• SUCCESS
• WARNING
• FAILED
4. Click OK to add the new output.
To modify an existing output:
1. Select the desired output in the list and click the Edit button, or double-click the desired output in the list. The Output Properties dialog opens, as shown in Figure 2-2.
2. Make the desired changes to the output parameters. Please note that you are not allowed to modify the name of a predefined output.
3. Click OK to save the settings.
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Figure 2-2. Output Properties
To remove an existing output:
1. Select the output to remove.
2. Click the Delete button. The output is deleted from the list.
To save your Test Library Settings and Outputs configuration to a file:
1. Click the Save Library Settings button. The Save As dialog opens.
2. Choose the location and type in the file name, then click the Save button. The library settings are saved to a file with the .lsf extension (Library Settings File).
To load an existing configuration for the test libraries and outputs:
1. Click the Load Library Settings button. The Open dialog opens.
2. Browse for the file where the desired execution settings are stored (an .lsf file) and open it. The settings are applied to your configuration.
Configuring Scenario Editor Default Settings
Choose Scenario Editor Defaults in the left pane of the Global Settings window to define the default parameters for each SIP, Skinny, H.323, MGCP, H.248/MEGACO, Digital T1/E1, and RTP test library function (Figure 2-3).
Note: The predefined outputs cannot be removed.
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Figure 2-3. Scenario Editor Defaults
The function parameter configuration can be saved to a file, which enables you to reapply the configuration at a later time.
To save the default parameters values to a file:
1. Click the Save Defaults button. A Save As dialog opens.
2. Choose the location and type in the file name, then click the Save button. Please note that the file has the .dft extension (Defaults Settings File).
To load the default values from a file:
1. Click the Load Defaults button. An Open dialog opens.
2. Browse for the file where the desired execution settings are stored (a .dft file) and open it. The settings are applied to your configuration.
To restore the default values for a function, select the function in the left tree pane and click the Restore Defaults button.
Note: When placing a script function in the Scenario Editor, its parameters are those set in the Scenario Editor Default window. These default parameters can be changed at any time by double-clicking the function body and editing as desired.
The parameters available for each script function are described in detail in Voice Functions Reference on page 3-1.
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3Chapter 3: Voice Functions Reference
This section describes the script functions in the Voice Test Libraries.
• VoIP SIP Functions Library on page 3-1.
• VoIP Skinny Functions Library on page 3-41.
• VoIP Media Functions Library on page 3-86.
• VoIP Flow Functions Library on page 3-111
• VoIP H323 RAS Library on page 3-116
• VoIP H323 Functions Library on page 3-118
• VoIP H248 Functions Library on page 3-123
• VoIP MGCP Functions Library on page 3-138
• Digital T1/E1 Functions Library on page 3-155
VoIP SIP Functions Library
The VoIP SIP Test Library contains the following script functions:
• Send Request on page 3-2.
• Send Response on page 3-9.
• Wait Request on page 3-13.
• Wait Response on page 3-16.
• Wait Message on page 3-19.
• Retransmit Last Message on page 3-27.
• Extract Variables on page 3-27.
• T.38 Fax Session on page 3-100
• MSRP Session on page 3-33
• MSRP Control on page 3-38
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Send Request Sends a SIP request.
Send Request Properties: Parameters
Table 3-1 describes the Send Request function parameters.
Table 3-1. Send Request - Parameters
Name Description
SIP Message The SIP message to transmit. It can be edited manually in the message preview window, imported from a text file, or generated using the Template GUI.
Load from file Loads the SIP message to transmit from an existing text file.
Create From Template
Generates a correct SIP message based on existing message templates, as described in Creating a SIP Message from Template on page A-1.
Edit Options
Case Sensitive
Enables/disables case-sensitivity for the SIP message content. By default, case-sensitivity is disabled.
Change Case
If enabled, the SIP method names are written in uppercase letters (and in blue color) in the SIP message. This option works only if the Case Sensitive option is disabled. By default, it is enabled.
Available SIP methods are INVITE, ACK, BYE, CANCEL, OPTIONS, REGISTER, MESSAGE, NOTIFY, SUBSCRIBE, REFER, PRACK, INFO, UPDATE.
Delay Before Execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
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Message Body Defines the SDP information sent in the SIP message body:Send Audio SDP – Enables/disables sending audio SDP information. When this option is selected, the SDP can be transmitted as:
• Offer – (default) Sends the SDP information as defined in the test/activity configuration by the VoIP activity codecs list in the Codec Settings page.
• Negotiated – Sends the SDP information as negoti-ated between the SIP endpoints.
• Hold Session – Sends the SDP information such as to establish a hold session.
Send Custom Message Body – Enables/disables sending custom messages.
Clicking the Edit Custom opens the Custom Message Body window, which enables you to define the message body by freely editing it, or based on a predefined template. You can also add multipart message bodies, as described in Support for Multipart SIP Messages on page E-1.
The following options are available in the Custom Message Body window:
• Evaluate Expression between character(s) – the expression between the user-defined characters is evaluated.
• Extract SDP Information – the custom body is parsed by the SDP parser to extract SDP Offer/Answer media streaming information. Selecting this option implies updating, adding, or deleting media streams for the current endpoint; otherwise, the content body is ignored from the SDP Offer/Answer exchange point of view.
• Send unmodified SDP in case of negotiation – Sends the complete capabilities information in case of nego-tiation.
• Show CR/LF – Shows the CR/LF character in the SIP message body.
• Create from template – Creates a custom message body based on one of the following predefined tem-plates: Simple Session, Simple Secure Session, Multiple Sessions, G723 at 5.3 Bitrate, G729 Annex A, AMR Octet Aligned.
Clicking the Variables enables you to access scenario variables for use in the SIP custom message body.
Note: In order for the media to be correctly set up, the $VOIP_MediaIP and $VOIP_MediaBasePort scenario variables are used in the custom message body.
Table 3-1. Send Request - Parameters (Continued)
Name Description
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Send Request Properties: Behavior
This page allows you to specify the destination for the SIP message, as well as the parameter values, the transport protocol to use, and the dialog layer (new/existing dialog). Table 3-2 describes the Send Request Behavior parameters.
Note: Fields highlighted blue (shown in the image below) in the script function configuration tabs indicate that expressions using scenario variables and numerical values are accepted as input in these fields.
Table 3-2. Send Request Properties - Behavior Parameters
Parameter Description
Auto-Variables
Allows you to set the values for the SIP auto variables.
Custom Behavior
If selected, allows you to override the destination for the SIP message by specifying a destination address and port.
NOTE: If this option is enabled, all the other settings in the SIP message are ignored.
Transport Layer
Sets the function-level transport protocol to use. You have the following options:
• TCP
• UDP
• Last sent transport type – It uses the same protocol used by the last sent message. If it cannot be evaluated, the default is UDP.
• Last received transport type – It uses the same protocol used by the last received message. If it cannot be evaluated, the default is UDP.
Dialog Layer Specifies if the script function generates a new SIP dialog or uses the existing dialog. In SIP terms, a dialog is a SIP communication established between two UAs that persists for some time. It is established by SIP messages such as a 2xx response to an INVITE request, and is identified by a call identifier, a local tag, and a remote tag.
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Send Request Properties: Flow Manager
By overriding activity-level settings such as destination or transfer addresses, the settings in this page control the way the SIP message is transmitted. Available options are described in Table 3-3.
Remove previously received messages
If selected, the message queue is emptied at function execution time. When the Global option is selected, the value from the Global Settings>Library Settings and Outputs>VoIP>SIP page is used.
Extended variables support
If selected, enables support for test scenario variables, both read-only and custom, used in SIP message headers. With variable support enabled, at function execution time scenario variables are evaluated and replaced with the actual values.
Note: This option does not affect auto variables.
Table 3-3. Send Request Properties - Flow Manager
Parameter Description
Send this message based on
Defines the transmission mode, using either configured activity-level settings or overriding settings specified in this page.
Default Settings The message is transmitted using the predefined activity settings.
Redirect Information
The message is transmitted using redirect information included in the message. Current implementation limits the number of redirect information to 3 addresses, referenced as First, Second, or Third address.
If there is no redirect info available, the function returns an error message. If an IP address is specified exceeding the number of redirect info items (for example, you choose Third Address and only 2 redirects are defined), the First Address is automatically selected.
Table 3-2. Send Request Properties - Behavior Parameters (Continued)
Parameter Description
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Raw Information The message is transmitted as is, with the auto variables not being evaluated. Available options are:
• Using Test/Activity settings – The mes-sage uses the activity settings.
• Using DNS or IP address – The mes-sage uses the specified IP address or DNS name and port (mandatory). If the DNS name is used, the application checks it using the current network set-tings on the local machine. A warning message displays if the address cannot be resolved on the network.
Evaluate the expressions in the proper fields – If enabled, expression(s) contained in any of the above fields are evaluated.
Semantic Information
You can choose between:
• Using Test/activity settings – It uses the settings in the activity configuration.
• Using DNS or IP address – It uses the specified IP address or DNS name and port (mandatory). If the DNS name is used, the application checks it using the current network settings on the local machine. A warning message displays in case the address cannot be resolved on the network. When choosing this option, a proxy or registrar role can be selected for the IP address.
You can override the activity level Desti-nation Phone, Destination IPAddress, Destination Port, and TransferAddress settings by using variables or expres-sions in the fields below.
When the activity is contained in a VoIPSIP cloud, you can also overrride the activity level Source Address and Source Port, default values being the address and port of the first SIP Proxy Server that is part of the cloud.
Evaluate the expressions in the proper fields – If enabled, expression(s) contained in any of the above fields are evaluated.
Retransmis-sion Settings
Specifies retransmission settings that override the activity level settings defined in the VoIP Plug-in SIP Settings tab.
Table 3-3. Send Request Properties - Flow Manager (Continued)
Parameter Description
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Follow settings specified at activity level
When selected, the activity-level retransmission settings are used. You can, however, specify different T1 and T2 timer values by entering values in the fields below.
Disable retransmission
When selected, the message is not retransmitted, regardless of the activity-level retransmission settings.
Table 3-3. Send Request Properties - Flow Manager (Continued)
Parameter Description
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Send Request Properties: Extract Variables
This page allows you to select the SIP variables you want to be extracted from the SIP message.
Send Request Properties: Authentication
Available authentication settings are described in Table 3-4.
Note: The SIP test library also comprises a standalone script function that performs variable extraction, as described in Extract Variables on page 3-27.
Table 3-4. Send Request Properties - Authentication
Parameter Description
Override UAC Side Authentication
If enabled, the authentication credentials are overwritten with information specified in the fields described below:
• Realm
• User Name
• Password
These fields accept string values (written between quotation marks) and variables (prefixed by the ‘$’ character).
Digest Preferred QOP: Specifies the quality of protection level to use when the server offers multiple choices. Available choices are:
• auth(entication)
• auth(entication)-int(egrity)
• global (default)
When the Global option is selected, the value from the Global Settings>Library Settings and Outputs>VoIP>SIP page is used.
Note: The auth-int setting may lead to possible problems when traversing Network Address Translators (NATs), Back-to-Back UAs (B2BUAs) and Application Level Gateways (ALGs), any of which may modify the body in order to permit the SIP request to traverse some form of network boundary. In this case, the NAT/B2BUA/ALG must also act as an endpoint and police and possibly modify the authentication header.
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Send Request Properties: Outputs
Table 3-5 describes the outputs available for the Send Request function.
Send Response Sends a SIP response.
Send Response Parameters
Table 3-6 describes the Send Response function parameters.
Table 3-5. Send Request Properties - Outputs
Output Name
Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Transport Failure
The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
Table 3-6. Send Response - Parameters
Name Description
SIP Message The text SIP message to transmit. It can be edited manually in the message preview window, imported from a text file, or generated using the Create from Template GUI.
Load From File Loads from a text file the SIP message to transmit.
Create From Template
Generates the SIP message based on existing message templates. For more details on this window enabling you to create a correct SIP message, refer to Creating a SIP Message from Template on page A-1.
Edit Options
Case Sensitive
Enables or disables case sensitivity of the SIP message.
Change Case
If enabled, the SIP method names are written in uppercase letters (and in blue color) in the SIP message. This option works only if the Case Sensitive option is disabled. By default, it is enabled.
Available SIP methods are INVITE, ACK, BYE, CANCEL, OPTIONS, REGISTER, MESSAGE, NOTIFY, SUBSCRIBE, REFER, PRACK, INFO, UPDATE.
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Delay Before Execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Message Body Defines the SDP information sent in the SIP message body:
Send Audio SDP – Enables/disables sending audio SDP information. When this option is selected, the SDP can be transmitted as:
• Offer – (default) Sends the SDP as defined in the test/activity configuration by the VoIP activity codecs list in the Codec Settings page.
• Negotiated – Sends the SDP information negotiated between two SIP endpoints.
• Hold Session – Sends SDP such as to establish a hold session.
Send Custom Message Body – Enables/disables sending custom messages. You can also add multipart message bodies, as described in Support for Multipart SIP Messages on page E-1.
Clicking the Edit Custom opens the Custom Message Body window, which enables you to define the message body by freely editing it, or based on a predefined template.
Table 3-6. Send Response - Parameters (Continued)
Name Description
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Send Response Properties: Behavior
This section allows you to specify the destination of the SIP message, as well as the parameter values and the transport protocol to use. Table 3-7 describes the Send Response Behavior parameters.
The following options are available in the Custom Message Body window:
• Evaluate expression between character(s) – the expression between the user-defined characters is being evaluated.
• Extract SDP Information – the custom body is parsed by the SDP parser to extract SDP Offer/Answer media streaming information. Selecting this option implies updating, adding, or deleting media streams for the cur-rent endpoint; otherwise, the content body is ignored from the SDP Offer/Answer exchange point of view.
• Send unmodified SDP in case of negotiation: Sends the complete capabilities information in case of negoti-ation.
• Show CR/LF – Shows the CR/LF character in the SIP message body.
• Create from template – Creates a custom message body based on one of the following predefined tem-plates: Simple Session, Simple Secure Session, Multiple Sessions, G723 at 5.3 Bitrate, G729 Annex A, AMR Octet Aligned.
Clicking the Variables enables you to access scenario variables for use in the SIP custom message body.
Table 3-7. Send Response Properties - Behavior Parameters
Parameter Description
Auto-Variables Allows you to set the values for the SIP auto variables.
Custom Behavior Allows you to specify the destination of the SIP message using custom value for destination address and destination port.
Transport Layer Sets the function-level transport protocol to use:
• TCP
• UDP
• Last sent transport type – It uses the same protocol used by the last sent message. If it cannot be evaluated, the default is UDP.
• Last received transport type - It uses the same protocol used by the last received message. If it cannot be evalu-ated, the default is UDP.
Table 3-6. Send Response - Parameters (Continued)
Name Description
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Send Response Properties: Flow Manager
The settings in this page control the way the SIP message is transmitted. Avail-able options are described in Table 3-8.
Dialog Layer Specifies if the script function uses the existing dialog or creates a new one. A dialog is a peer-to-peer SIP relationship between two UAs that persists for some time. A dialog is established by SIP messages, such as a 2xx response to an INVITE request, and is identified by a call identifier, a local tag, and a remote tag.
Remove previously received messages
If selected, the message queue is emptied at function execution time. When the Global option is selected, the value from the Global Settings | Library Settings and Outputs | SIP page is used.
Extended variables support
If selected, enables support for test scenario variables, both read-only and custom, used in SIP message headers. With variable support enabled, at function execution time scenario variables are evaluated and replaced with the actual values.
Note: This option does not affect auto variables.
Table 3-8. Send Response Properties - Flow Manager
Parameter Description
Send this message based on
Defines the message transmission mode, using either configured activity-level settings or overriding settings specified in this page.
Default Settings The message is transmitted with the default activity settings (default).
Raw Information The message is transmitted as is, with the auto variables not being evaluated. Available options are:
• Using Test/Activity settings – The message uses the activity settings.
• Using DNS or IP address – The message uses the specified IP address or DNS name and port (mandatory). If the DNS name is used, the application checks it using the current network settings on the local machine. A warning message displays if the address cannot be resolved on the network.
Evaluate the expressions in the proper fields – If enabled, expression(s) contained in any of the above fields are evaluated.
Table 3-7. Send Response Properties - Behavior Parameters
Parameter Description
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Send Response Properties: Extract Variables
This section allows you to select the SIP variables you want to be extracted from the SIP message.
Send Response Properties: Outputs
For more information, please refer to Send Request Properties: Outputs on page 3-9.
Wait Request Waits for a SIP request.
Wait Request Parameters
Table 3-9 on page 3-13 describes the Wait Request function parameters.
Retransmission Settings
Specifies retransmission settings that override the activity level settings defined in the VoIP Plug-in SIP Settings tab.
Follow settings specified at activity level
When selected, the activity-level retransmission settings are used. You can, however, specify different T1 and T2 timer values by entering values in the fields below.
Disable retransmission
When selected, the message is not retransmitted, regardless of the activity-level retransmission settings.
Note: The SIP test library also comprises a standalone script function that performs variable extraction, as described in Extract Variables on page 3-27.
Table 3-8. Send Response Properties - Flow Manager
Parameter Description
Table 3-9. Wait Request - Parameters
Name Description
SIP Message The SIP request for which to wait, which can be generated using the Create from Template GUI.
Create From Template
Generates the SIP message based on existing message templates. For details on this window enabling you to create a correct SIP message refer to Creating a SIP Message from Template on page A-1.
Ignore SDP If selected, the message body is ignored from the SDP offer/answer exchange point of view.
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Wait Request Properties: Extract Variables
This section allows you to select the SIP variables that you want to be extracted from the SIP message.
Wait Request Properties: Retransmission
Table 3-10 describes the retransmission settings available for the Wait Request function.
Extended Variables Support
If selected, enables support for test scenario variables, both read-only and custom, used in SIP message headers. With variable support enabled, at function execution time scenario variables are evaluated and replaced with the actual values.
Note: This option does not affect auto variables.
Delay Before Execution
Delays the function execution by a duration that can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, Sleep 2000 in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
Timeout The time (in ms) the function waits for the received message to match the template. If this time period terminates without having a match, the function enables the Timeout output. It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Delay Between Digits, in milliseconds (ms)
• PHONE_WAIT_TIME, in milliseconds (ms)
• MGCP Timeout, in milliseconds (ms)
Remove previously received messages
If selected, the message queue is emptied at function execution time. When the Global option is selected, the value from the Global Settings>Library Settings and Outputs>VoIP>SIP page is used.
Note: The SIP test library also comprises a standalone script function that performs variable extraction, as described in Extract Variables on page 3-27.
Table 3-9. Wait Request - Parameters (Continued)
Name Description
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Wait Request Properties: Outputs
Table 3-11 describes the outputs available for the Wait Request function.
Table 3-10. Wait Request Properties - Retransmission
Parameter Description
Retransmission Settings
When a request message comprised in a SIP transaction is matched, these settings define the conditions for ending or keeping transaction-level retransmissions of messages.
Note: For each script function that has retransmissions enabled and carries a SIP message subject to retransmissions, a retransmission rule is created. Each scenario channel maintains a list of known retransmission rules, one for every such script function that started retransmissions.
Clean matched retransmission rule
If selected, the matched retransmission rule is stopped.
Keep selected retransmission rules active (and clean the other)
If selected, from all known retransmission rules displayed in the Started in Function drop-down, only the selected rule(s) remain(s) active.
Note: For example, considering the transaction messages flow shown in the image below:
whereby an INVITE message is sent by endpoint A, and endpoint B retransmits a response 200 OK message for a number of times at increasing intervals.
Considering the retransmission settings for the Wait Request (ACK) script function executed by endpoint B, the following behaviors can be implemented depending on the retransmission settings:
• If the Clean matched retransmission rule option is selected, the 200 OK message retransmission is stopped after receiving the ACK request.
• If the Keep selected retransmission rule active option is selected and the Send 200 OK rule is checked, the 200 OK message is retransmitted after receiving the ACK request, until the retransmission timeout expires.
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Wait Response Waits for a SIP response.
Wait Response Parameters
Table 3-12 describes the Wait Response function parameters.
Table 3-11. Wait Request Properties - Outputs
Output Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Timeout This output is enabled if the request is not received within the specified timeout.
The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
Table 3-12. Wait Response - Parameters
Name Description
SIP Message The SIP response for which to wait. It can be generated using the Create from Template GUI.
Create From Template
Generate the SIP message based on existing message templates. For more details on this dialog that helps you create a correct SIP message, refer to Creating a SIP Message from Template on page A-1.
Ignore SDP If selected, the message body is ignored from the SDP offer/answer exchange point of view.
Extended Variables Support
If selected, enables support for test scenario variables, both read-only and custom, used in SIP message headers. With variable support enabled, at function execution time scenario variables are evaluated and replaced with the actual values.
Note: This option does not affect auto variables.
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Wait Response Properties: Extract Variables
This page allows you to select the SIP variables you want to be extracted from the SIP message.
Wait Response Properties: Retransmission
Table 3-13 describes the retransmission settings available for the Wait Response function.
Delay Before Execution
Delays the function execution by a duration that can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, Sleep 2000 in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
Timeout The time (in ms) the function waits for the received message to match the template. If this time period terminates without having a match, the function enables the Timeout output. It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Delay Between Digits, in milliseconds (ms)
• PHONE_WAIT_TIME, in milliseconds (ms)
• MGCP Timeout, in milliseconds (ms)
Table 3-12. Wait Response - Parameters (Continued)
Name Description
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Table 3-13. Wait Response Properties - Retransmission
Parameter Description
Retransmission Settings
When a request message comprised in a SIP transaction is matched, these settings define the conditions for ending or keeping transaction-level retransmissions of messages.
Note: For each script function that has retransmissions enabled and carries a SIP message subject to retransmissions, a retransmission rule is created. Each scenario channel maintains a list of known retransmission rules, one for every such script function that started retransmissions.
Clean matched retransmission rule
If selected, the matched retransmission rule is stopped.
Keep selected retransmission rules active (and clean the other)
If selected, from all known retransmission rules displayed in the Started in Function drop-down, only the selected rule(s) remain(s) active.
Note: For example, considering the transaction messages flow shown in the image below:
whereby an INVITE message is sent repeatedly for a number of times (retransmitted) by endpoint A, and endpoint B transmits a response 200 OK message.
Considering the retransmission settings for the Wait Response (200 OK) script function executed by endpoint A, the following behaviors can be implemented:
• If the Clean matched retransmission rule option is selected, the INVITE message retransmission is stopped after receiving the 200 OK response.
• If the Keep selected retransmission rule active option is selected and the Send INVITE rule is checked, the INVITE message is further retransmitted after receiving the 200 OK response, until the retransmission timeout expires.
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Wait Response Properties: Outputs
For further information, please refer to Wait Request Properties - Outputs on page 3-16.
Wait Message Waits for a SIP request or response message and matches it against one of the message templates defined in the script function.
Wait Message Properties: Templates
The received message is compared with each template from a list of up to 10 templates, starting with the template at the top of the list.
Comparison of the received message with a message template is based on:
• The template type (request, response)
• Request options: Request Line Method (INVITE, ACK, OPTIONS,…)
• Response options: Status Code Value (200 (OK), 182, …)
• For each template, a logical scheme may be defined at header level (Call-Id, Contact, CSeq, From, To). If a header is not checked, the Wait Message func-tion does not take it into consideration.
In case of a match, the function enables the output corresponding to the matched template, or a Timeout output if no match was found. The Wait Message avail-able options are listed in Table 3-14.
Note: The Wait Message function adds flexibility to scenario configuration, by enabling you to receive and match multiple messages using a single script function, instead of multiple Wait Request / Wait Response ones.
Table 3-14. Wait Message Properties - Templates
Option Description
Template Name User-defined template name in the list. By default, names with the msg_name#01, msg_name#02, ….format are assigned.
Template Type The type of message for which to wait. Available options are:
- (request)
- (response)
For more details about these message types, please refer to The Template Window on page 3-20.
New TemplateCreates a new template, configured using the Template window, as described in The Template Window on page 3-20.
Delete Template
Deletes the selected templates (supports multiple selections).
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The Template Window
Table 3-15 describes the Template window parameters for the Wait Message function.
Move UpMoves the selected templates one position up (supports multiple selections).
Move DownMoves the selected templates one position down (supports multiple selections).
Note: Moving up/down the headers establishes the order in which they are matched by the Wait functions with the defined templates.
PreviewOpens the preview window that displays the SIP message headers for the selected message.
Note: To modify an existing message template, double-click the template to open the Template editor window.
Wait for any message
If enabled, the Wait Message function exposes an additional output, Other, that is enabled if the received message does not match any of the defined SIP message templates.
When this option is disabled, the function enables the Timeout output if the received message does not match any of the defined SIP message templates (default).
Note: This option can be selected only if there is at least one template defined in the Templates list.
Message headers The general headers included with the selected template. For more details, please refer to Message Header Parameters on page 3-21.
Ignore SDP If selected, the message body is ignored from the SDP offer/answer exchange point of view.
Observations: User-defined field with a comment referring to the selected template, which does not influence the function behavior.
Restore Defaults Sets the function parameters to their default settings, as specified for the Wait Message function in the Global Settings> Scenario Editor Defaults>VoIP>Wait Message page.
OK Applies the settings and closes the Properties window.
Cancel Discards the changes and closes the Properties window.
Apply Applies the settings.
Table 3-14. Wait Message Properties - Templates (Continued)
Option Description
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Message Header Parameters
Once a template has been added to the list, message header parameters must be edited according to the following conventions for the Value column content:
• |ANY | – the corresponding parameter must be present and may take any value. This is the matching condition for the parameter in the received mes-sage.
• |NOT_ALLOWED| – the corresponding parameter must not be present. This is the matching condition for the parameter in the received message.
• n/a – the corresponding parameter has no available options/values.
• User-defined field – this is the comment for the parameters that support user-defined values. The default value is also specified.
The other lines in the Value column are SIP options. For more details, please refer to RFC 3261 (SIP).
Table 3-16 describes the Message Headers parameters.
Table 3-15. Wait Message Properties - Template Options
Option Description / Available Values
Template Name User-defined template name. By default, names with msg_name#01, msg_name#02,…formats are assigned.
Message Type: Type of SIP message for which to wait. Available types are:
- Request (default)
- Response
Request options Request Line Method
The message template for which to wait. Available options are:
• INVITE
• ACK
• OPTIONS
• BYE
• CANCEL
• REGISTER
• REFER
• NOTIFY
• SUBSCRIBE
• MESSAGE
• REFER
• PRACK
• INFO
• UPDATE
Response options Status Code – the status code for which to wait.
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Table 3-16. Message Headers - Parameters
Header / Parameters Available Options / Values
Call-ID Call-ID
i
|ANY|
|NOT_ALLOWED|
Call-Id User-defined field. The default is word@word.
Contact Contact
m
|ANY|
|NOT_ALLOWED|
STAR n/a
contact-param n/a
name-addr |ANY_NAME_ADDR|
|NOT_ALLOWED_NAME_ADDR|
addr-spec |ANY_ADDR_SPEC|
|NOT_ALLOWED_ADDR_SPEC|
Param Q
expires
|ANY_PARAM|
|NOT_ALLOWED_PARAM|
Value User-defined field. The default is an empty string.
Cseq Cseq
|ANY|
|NOT_ALLOWED|
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Numeric Value User-defined field. The default is 0.
Method INVITE
ACK
OPTIONS
BYE
CANCEL
REGISTER
REFER
NOTIFY
SUBSCRIBE
MESSAGE
PRACK
INFO
UPDATE
The default is an empty string.
From From
f
|ANY|
|NOT_ALLOWED|
name-addr |ANY_NAME_ADDR|
|NOT_ALLOWED_NAME_ADDR|
addr-spec |ANY_ADDR_SPEC|
|NOT_ALLOWED_ADDR_SPEC|
Param Tag
|ANY_PARAM|
|NOT_ALLOWED_PARAM|
Value User-defined field. The default is an empty string.
To To
t
|ANY|
|NOT_ALLOWED|
name-addr |ANY_NAME_ADDR|
|NOT_ALLOWED_NAME_ADDR|
addr-spec |ANY_ADDR_SPEC|
|NOT_ALLOWED_ADDR_SPEC|
Param Tag
|ANY_PARAM|
|NOT_ALLOWED_PARAM|
Table 3-16. Message Headers - Parameters (Continued)
Header / Parameters Available Options / Values
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Checks the header(s) that you want to be taken into consideration. Uncheck the header(s) that you do not want to be taken into consideration.
Wait Message Properties: Params
Table 3-17 describes the Wait Message Properties parameters.
Value User-defined field. The default is an empty string.
Table 3-17. Wait Message Properties - Parameters
Name Description
Extend Variables Support
If selected, enables support for scenario variables—both read-only and custom—used in SIP message headers. With variable support enabled, at function execution time, scenario variables are evaluated and replaced with the actual values.
Note: This option does not affect auto variables.
Delay Before Execution
Delays the function execution by a duration that can be specified as:
• Static Expression – Fixed Delay Before Execution dura-tion. It can be used for synchronization reasons. It can be a value in milliseconds (ms), or a formula.
• Random Between Expressions – Random Delay Before Execution duration value in the specified interval. It can be used to simulate real-life conditions.
• A user-defined delay (VoIP constant), chosen from those available in the Resource Pool | VoIP Constants.
The default is Static Expression, 0 ms.
Timeout The time (in ms) the function waits for a specific incoming message defined by templates. If this time period terminates without matching any of the defined message templates, the function exits on the Timeout output. It can be specified as:
• Static expression/value, in milliseconds (ms)
• Random between two values, in milliseconds (ms)
• A user-defined timeout (VoIP constant), chosen from those available in the Resource Pool>VoIP Constants.
The default is Static Expression, 20000 ms.
Remove previously received messages
If selected, the message queue is emptied at function execution time. When the Global option is selected, the value from the Global Settings>Library Settings and Outputs>VoIP>SIP page is used.
Table 3-16. Message Headers - Parameters (Continued)
Header / Parameters Available Options / Values
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Wait Message Properties: Extract Variables
This page allows you to select the SIP variables you want to be extracted from the SIP message.
Wait Message Properties: Retransmission
Table 3-18 describes the outputs available for the Wait Response function.
Table 3-18. Wait Message Properties - Retransmission
Parameter Description
Retransmis-sion Settings
When a message comprised in a SIP transaction is matched, these settings define the conditions for ending or keeping transaction-level retransmissions of messages.
Note: For each script function that has retransmissions enabled and carries a SIP message subject to retransmissions, a retransmission rule is created. Each scenario channel maintains a list of known retransmission rules, one for every such script function that started retransmissions.
Clean matched retransmission rule
If selected, the matched retransmission rule is stopped.
Keep selected retransmission rules active (and clean the other)
If selected, from all known retransmission rules displayed in the Started in Function drop-down, only the selected rule(s) remain(s) active.
From Template Since the Wait Message function matches multiple message, the drop-down enables you to specify the list of retransmission rules kept active for each of the expected message templates.
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Wait Message Properties: Outputs
Table 3-19 describes the outputs available for Wait Message function.
Note: For example, considering the messages flow shown in the image below:
whereby an INVITE message is retransmitted repeatedly by endpoint A at increasing intervals, and endpoint B responds with the successive 100 Trying, 180 Ringing and the 200 OK messages. For added flexibility, all response messages can be received using a single Wait Message script function, instead of multiple Wait Response ones, as shown in the image below:
The flow above corresponds to the following script functions settings:
• For the 100 Trying template, the Keep selected retransmission rule active option is selected and the Send Invite rule is enabled
• For the 180 Ringing template, the Keep selected retransmission rule active option is selected and the Send Invite rule is enabled
• For the 200 OK template, the Keep selected retransmission rule active option is selected, without any retransmission rule checked.
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Retransmit Last Message
Re-sends the last transmitted SIP Message.
Retransmit Last Message: Outputs
For more information, please refer to Send Request Properties: Outputs on page 3-9.
Extract Variables Performs the extraction into one or more variables from a SIP message, or from parts of a SIP message.
Variable extraction is based on defining the extraction rules, as described in Extraction Rules Definition on page 3-28. Existing variable extraction rules can be exported to the Resource Pool of the VoIP plug-in for later use; reuse of extraction rules from the Resource Pool is done by importing.
Extract Variables: Parameters
Displays a list of variables selected for extraction and enables you to define the variable-specific extraction rules using a three-step wizard.
Variable Operations
Before defining the extraction rules, create a variable by clicking the and define it as:
Table 3-19. Wait Message Properties - Outputs
Output Name Description
The outputs corresponding to message template (max 10)
If the received SIP message matches one template, the corresponding output is enabled.
The default resolution for this output is SUCCESS.
Timeout This output is enabled if the received SIP message does not match any of the message templates in the specified timeout.
The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
NOTE: Any of the outputs corresponding to message template availability in this dialog depends on how the function options are configured.
Note: Since extraction of variables is done by a SIP parser software component, extraction rules definition must be compliant with the SIP syntax requirements.
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• A Temporary variable – can be used only in the body of this function.
• A Scenario variable – is a user-defined, local variable.
In addition to creating a variable, further operations that can be performed on variables are:
• Editing a variable, done by clicking the .
• Deleting a variable, done by clicking the .
• Reordering the variable sequence, done by clicking the and s.
Extraction Rules Definition
Extraction rules are defined using a wizard that comprises the following steps:
1. When and Where: Specifies a condition upon which extraction occurs and enables you to specify the SIP message from which the variable is extracted.
2. What: Defines the variable extraction scope, that is, if the variable is extracted from the entire message, or from parts of it.
3. Refine: Defines further processing operations that can be performed on the extracted variable string.
The following is a description of the options available for each extraction step listed above.
When and Where Step Parameters
Table 3-20 lists the parameters available for the When and Where step.
Note: Once a variable extraction rule has been defined, you can export it by
clicking the . To import an existing rule, click the .
Table 3-20. When and Where Parameters
Area, Option Description
When Search only when the following expression is true
Selecting this option enables you to type in a conditional expression that determines whether variable extraction is done or not.
For a description of the accepted elements and syntax, refer to The Expression Evaluator Syntax on page B-1.
Where Search in last transmitted message
Select this option to perform the variable extraction from the last transmitted message.
Search in last received and matched message
Select this option to perform the variable extraction from the last received and matched message.
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What Step Parameters
Table 3-21 lists the parameters available for the What step.
Search in variable Select this option to perform the variable extraction from another variable that is specified in the adjacent drop-down control.
Table 3-21. What Parameters
Parameter Description
Entire SIP message Select this option to extract the variable from the entire SIP message.
First line Select this option to extract the variable from the first line, or from parts of it, both for request and response messages. The following options are available:
• Entire First Line
• Request Line - Method
• Request Line - Request-URI
• Request Line - SIP Version
• Status Line - SIP Version
• Status Line - Status-Code
• Status Line - Reason Phrase.
Keep last CR/LF: Keeps the last Carriage Return/Line Feed character extracted into the variable.
Table 3-20. When and Where Parameters
Area, Option Description
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Header Select this option to extract the variable from one or multiple occurrences of a message header. When choosing this option, the following options become available:
• Header type: Selects the header type from which the variable is extracted.
• Compact form: After you have selected a header type, this field is automatically filled in with the header’s compact form.
• Occurrences: Sets the range of extracted occurren-ces of the chosen header type. If the occurrence count is greater than 1, multiple message lines cor-responding to the multiple header rows are pro-cessed and extracted into the variable.
Further extract options are:
• The whole header value: Extracts the whole line, except for the header name.
• Extract also the header name: Extracts the whole line, including the header name.
• Header value without parameters: Extracts the header value, without any parameters. This option is available only for occurrence counts equal to 1.
• Value of parameter named: Extracts the value of the named parameter. This option is available only for occurrence counts equal to 1.
• Extract headers in reverse order: When the resulting occurrences count is higher than 1, choosing this option processes the occurrences in reverse order, starting from the last up to the first.
• Keep last CR/LF: Keeps the last Carriage Return/Line Feed character extracted into the variable.
For example, when extracting multiple whole SIP headers into a variable and subsequently parsing that variable in order to perform another extraction, selecting this option ensures that the SIP parser finds valid, CR/LF-terminated lines, as specified by the SIP grammar.
Message body • Entire Message Body: The entire message body is extracted, containing the SDP information and pos-sibly multiple parts. For information on multipart SIP messages see Support for Multipart SIP Messages on page E-1.
• SDP: Only the SDP portion of the message body is extracted.
Table 3-21. What Parameters (Continued)
Parameter Description
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Refine Step Parameters
Table 3-22 lists the parameters available for the Refine step.
Table 3-22. Refine Parameters
Parameter
Extract substring
Use delimiters Extracts a substring delimited by a start (Begins after parameter) and an end (Ends before parameter) string. Both the start and end strings are user-specified.
The Occurrence field defines the occurrence of the start and end strings.
Example: Considering that your variable is a string comprising several comma-separated values, you can choose:
• the ‘;’ character and an occurrence count of ‘2’ for the start string
• the ‘;’ character and an occurrence count of ‘3’ for the end string.
Then the extracted substring would be that located between the 2nd and the 3rd occurrence of the ‘;’ character.
Use position Extracts a substring delimited by a start (Extract from position parameter) and an end position (to position parameter).
Find & Replace
Enables you to perform replacement operations on the resulting variable using Find & Replace rules. A Find & Replace rule is
created by clicking the , which displays a dialog allowing you to specify the substring to search for, the replacement substring, and the occurrences for which to perform the replacement.
Further operations that can be performed on Find & Replace rules are:
• Click the to edit a rule.
• Click the to delete a rule.
• Click the and s to reorder the rules.
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Extract Variables: Outputs
Table 3-23 describes the outputs available for the Extract Variables function.
MSRP Send AUTH This function performs authentication for a MSRP endpoint against all relays specified in the VoIPSIP Peer activity’s MSRP tab. If the authentication process succeeds, the returned response code is 200 OK and the function exits on the OK output. If the authentication process does not complete successfully and a response is received other than 200 OK, if the received response is matched by any of the additional responses configured in the Rx Response tab, the function exits on the corresponding output, otherwise it exits on the Error output.
MSRP Send Auth: Tx Request Parameters
Table 3-24 describes the Send Auth Tx Request parameters.
MSRP Send Auth: Rx Response Parameters
Table 3-25 describes the Send Auth Rx Response parameters.
Table 3-23. Extract Variables - Outputs
Output Name
Description
Found The function completed successfully.
The default resolution for this output is SUCCESS.
Not Found The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
Table 3-24. Send Auth Tx Request Parameters
Name Description
Header name, Header value
Specifies the MSRP headers that are sent in the authentication request and enables you to configure them.
The From-Path and To-Path headers are always contained in requests and automatically are populated with VoIPSIP activity level settings (<AUTO>). Other headers (Expires, Authorization) can be included optionally and their value can be configured using automatic settings (<AUTO>) configured at the VoIPSIP activity level, or numeric values.
Override SIP authentication
If selected, this enables you to define other authentication settings (user name, password) than those specified in the SIP configuration page of the VoIPSIP activity.
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MSRP Send Auth: Output Settings
Table 3-26 describes the default outputs available for the Multimedia Session function. Note that for each additional configured response, a new output is created.
MSRP Session This function establishes an MSRP session for the simultaneous sending and receiving of text messages or files.
MSRP Session: Content Parameters
Table 3-27 describes the MSRP Session Content parameters.
Table 3-25. Send Auth Rx Response Parameters
Name Description
Response to be matched
Specifies the expected response from authentication requests (200 OK is default), or any other user-defined response. For each expected response message, a new script function output is created.
If the received response matches one of the specified responses, the script function exits on the output that corresponds to that response. If the response does not match any specified response, the function exists on the Error output.
To create a new expected response, click the
button, delete existing ones by clicking the
button.
For example, assuming you are expecting a 400 Bad request response during the authentication process, you can choose to match this response code by creating a 400 Bad request entry.
Table 3-26. Send Auth Output Settings
Output Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Timeout The function timed out while waiting for an authentication response.
The default resolution of this output is timeout.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
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MSRP Session: Settings
Table 3-28 describes the MSRP Session Settings parameters.
Table 3-27. MSRP Session Content Parameters
Name Description
Send MSRP Text Messages
Specifies a text to be sent across the MSRP session.
Send MSRP text messages: If selected, the specified text is sent.
Send this message: Specifies a number of times the message is sent.
Send each line in a separate message: If selected, each line (including the CRLF character) is sent as a separate message.
Send files through MSRP
When the Send Files through MSRP option is selected, these options specify the file to be sent across the MSRP session.
• Send synthetic file of size: A synthetically created file of the specified size is sent.
• Send custom file: A real, user-specified file with a maxi-mum size of 20 MB is sent.
• File negotiated in SDP: The transmitted file is one of the files defined in the VoIPSIP activity’s MSRP configu-ration page, and that is being negotiated by SDP, as stip-ulated by RFC5547 - A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer.
• File # from activity settings: The transmitted file is one specified in the VoIPSIP activity’s MSRP configuration page and identified by its table index.
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MSRP Session: Tx Requests Parameters
Table 3-29 describes the MSRP Session Tx Requests parameters.
Table 3-28. MSRP Session Settings
Name Description
Session control Specifies how the MSRP session is terminated.
• Finish the session when: This specifies an aggre-gated criterion (AND logical operator) for terminating the session:
• Finish the session when Tx is over: The MSRP session is terminated when the script function completes sending content.
• Finish the session when at least x messages have been received: If selected, the MSRP ses-sion is terminated when a specified number of messages have been received.
• Finish the session when file transfer Rx is over: If selected, the MSRP session is termi-nated when an incoming file transfer traffic is completed.
• Finish the session at the end of the call: If selected, the MSRP session is only terminated at the end of the SIP session.
Tx control Specifies how MSRP traffic is handled.
• Max Tx chunk size: If selected, the MSRP content is ‘chunked’ into portions of the specified size. The maximum value for the chunk size is 1 MB.
• Delay between text messages: If selected, the MSRP text content is send with delays of the speci-fied size between text messages.
Non-blocking execution
If selected, the MSRP Session script function starts executing and then returns control to the Scenario Editor, while continuing to execute in the background.
Note: When non-blocking execution is used, the script function’s output status can be determined using the MSRP Control script function.
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MSRP Session: Tx Responses Parameters
Table 3-30 describes the MSRP Session Tx Response parameters.
MSRP Session: CPIM Parameters
This tab adds support for the Message/CPIM MIME content type in MSRP (RFC 3862 and RFC 4975) and for the Instant Message Disposition Notifications (IMDN) mechanism (RFC 5438).
Table 3-31 describes the MSRP Session CPIM parameters.
Table 3-29. MSRP Session Tx Requests Parameters
Name Description
Send request structure
Specifies the structure of an MSRP Send request, as defined by RFC 4975, enabling you to select and configure MSRP header values.
The From-Path, To-Path, and Message-ID headers are always contained in requests and are automatically populated with VoIPSIP activity level settings (<AUTO> setting). Other headers can be included optionally and their values can be configured using automatic activity level settings (<AUTO>), predefined values, or custom values.
When clicking the Add Custom header button, one or more custom headers are added inside MSRP message, placed between the Failure-Report and the Content-ID header lines.
Note: IxLoad test scenario variables can be used as header values.
Table 3-30. MSRP Session Tx Responses Parameters
Name Description
Response Enables you to configure the response for received MSRP requests.
• Send automatic responses to all MSRP requests: If selected, responses are constructed automatically.
• Define custom response: If selected, the config-ured response (both response code and text com-ments can be configured) is sent for all received MSRP requests.
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Table 3-31. MSRP Session CPIM Parameters
Name Description
Use CPIM encapsulation
If selected, all MSRP SEND requests initiated by the current MSRP Session script function use a CPIM encapsulation.
The CPIM ‘To’ and ‘From’ headers will have the same value as the $VOIP_MSRP_CPIM_To and $VOIP_MSRP_CPIM_From IxLoad variables.
CPIM custom headers If selected, one or more custom headers are added inside the CPIM headers. The custom headers may contain VoIP variables.
Add disposition notification (IMDN)
If selected, IMDN notifications are requested for the received MSRP SEND requests. This mechanism is used together with CPIM.
The Positive delivery, Negative Delivery, Display notification options specify the requested notification types.
In the IxLoad implementation, an emulated VOIPSIP Peer activity can respond only with a positive delivery or a display notification.
Delay between notifications (ms)
The time interval that specifies the delay between the positive delivery of MSRP messages and the display notifications (if both were requested).
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MSRP Session: Output Parameters
Table 3-32 describes the outputs available for the MSRP Session function.
MSRP Control The MSRP Control function has the purpose of probing or controlling one (the previously executed) MSRP Session script function that had non-blocking behav-ior enabled.
Note: The following example illustrates an MSRP SEND request with CPIM and IMDN enabled:
MSRP 00000419da SEND
To-Path: msrp://ndc101cpm.npc.mobilephone.net:32398/n01s00i3t554BDFA5+121714;tcp
From-Path: msrp://ndc101cpm.npc.mobilephone.net:1024/db7b02;tcp
Message-ID: eb1c000400
Byte-Range: 1-303/303
Success-Report: no
Failure-Report: yes
Content-ID: <[email protected]>
Content-Description: A simple text message
Content-Type: message/cpim
From: <sip:[email protected]>
To: <tel:+18800305000>
DateTime: 2015-05-12T09:28:58Z
NS: imdn <urn:ietf:params:imdn>
imdn.Message-ID: 374b18a22290009433
imdn.Disposition-Notification: positive-delivery, display
Content-Type: text/plain
"Let’s communicate”
-------00000419da$
Table 3-32. MSRP Session Output Settings
Output Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
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If blocking execution is enabled for a previously executed MSRP Session func-tion, the output of the MSRP Control function indicates wheter the function exe-cution completed (MSRP Not Running output) or not (MSRP Running output).
MSRP Control: Control Action Parameters
Table 3-33 describes the MSRP Control Action parameters.
MSRP Control: Output Parameters
Table 3-34 describes the outputs available for the MSRP Control function.
Note: When running in non-blocking execution mode, script functions perform their tasks in background, allowing simultaneous handling of other signaling and media functions. A MSRP Session script function running in non-blocking mode always exits on the SUCCESS output, while the ‘true’ result of the function execution is obtained by evaluating the output of the MSRP Control function.
Table 3-33. MSRP Control Action Parameters
Name Description
Delay Before Execution
Delays the function execution by a duration that can be specified as:
• Static Expression – Fixed Delay Before Execution duration. It can be used for synchronization reasons. It can be a value in milliseconds (ms), or a formula.
• Random Between Expressions – Random Delay Before Execution duration value in the specified interval. It can be used to simulate real-life condi-tions.
The default is Static Expression, 100 ms.
Control Actions Specifies whether the function only probes the status of an executing MSRP Session function (Check for MSRP completion option), or whether it actually terminates it (Terminate MSRP option).
Table 3-34. MSRP Control Output Settings
Output Name Description
MSRP Running The previous MSRP function is still executing.
The default resolution for this output is SUCCESS.
MSRP Not Running
The previous MSRP function has completed execution.
The default resolution for this output is SUCCESS.
Timeout The function execution has generated a timeout.
The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
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VoIP Skinny Functions Library
The VoIP Skinny Test Library includes the following script functions:
• RegisterClient on page 3-41.
• UnregisterClient on page 3-45.
• OffHook on page 3-46.
• OnHook on page 3-48.
• NewCall on page 3-49.
• EndCall on page 3-50.
• MakeCall on page 3-51.
• WaitCall on page 3-52.
• AnswerCall on page 3-53.
• DialDigits on page 3-55.
• WaitDigits on page 3-56
• HoldCall on page 3-57.
• RetrieveCall on page 3-58.
• Setup XFER on page 3-59.
• Complete XFER on page 3-61.
• Transfer on page 3-63
• ForwardAllCalls on page 3-66.
• ParkCall on page 3-67.
• GetCallInfo on page 3-68.
• MeetMe on page 3-68
• RemoveLast ConferenceParty on page 3-70
• SendStimulus on page 3-71.
• SendSoftkey on page 3-72.
• IsSoftKeyAvailable on page 3-73.
• WaitForEvent on page 3-74.
RegisterClient Registers the Skinny client with a specified Cisco CallManager. If the client is already registered with the CallManager, this function is skipped.
Note: For tests comprising several loops, the registration state is maintained across loops in the test.
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The RegisterClient script function parameters are given in Table 3-35.
Table 3-35. RegisterClient Properties
Properties Description
Parameters Device Settings
Specifies the station settings for registration with the CallManager:
• Use Dial Plan: The registration name and type are those defined in the dial plan.
• Registration Name: An alternative registration name and type are provided.
Register to Enables you to specify a CallManager address and port overriding the settings from the Skinny Settings page:
• Server Address: The CallManager IP address
• Server Port: The CallManager port
Type The emulated station type, which can be one of the following:
• Device Station 30 SP+
• Device Station 12 SP+
• Device Station 12 SP
• Device Station 30 VIP
• Device Station Telecaster (Cisco IP Phone 7910)
• Device Station TelecasterMgr (Cisco IP Phone 7960)
• Device Station TelecasterBus (Cisco IP Phone 7940)
• Device Station Polycom (Cisco IP Phone 7935)
• Device Cisco IP Phone 7902
• Device Station VGC (Cisco VGC Phone)
Delay before execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
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Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Resource Pool | Constants category.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as for example Delay between digits or PHONE_WAIT_TIME
The default is Static Expression, 20000 ms.
Table 3-35. RegisterClient Properties (Continued)
Properties Description
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Advanced Permits you to specify a registration sequence with the CallManager. The registration sequence is given below:
Table 3-35. RegisterClient Properties (Continued)
Properties Description
Custom Skinny Registration Sequence: If selected, a custom sequence can be specified by clicking the message entries that are to be included in the custom sequence. When this option is not selected, the default sequence used ignores the StationKeepAliveAck and StationDisplayPromptStatus messages.
Messages Direction
StationKeepAlive To CCM
StationKeepAliveAck From CCM
StationAlarm To CCM
StationRegister To CCM
StationIpPort To CCM
StationRegisterAck From CCM
*StationHeadsetStatusMessage To CCM
StationCapabilitiesReq From CCM
StationCapabilitiesRes To CCM
StationTemplateReq To CCM
StationTemplateRes From CCM
*StationSoftKeyTemplateReq To CCM
*StationSoftKeyTemplateRes From CCM
*StationSoftKeySetReq To CCM
*StationSoftKeySetRes From CCM
StationLineStatReq To CCM
*StationSelectSoftKey From CCM
*StationDisplayPromptStatus From CCM
StationLineStat From CCM
StationSpeedDialStatReq To CCM
StationSpeedDialStat From CCM
*StationRegisterAvailableLinesMessages
To CCM
StationDateTimeReq To CCM
StationDefineDateTime From CCM
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UnregisterClient Unregisters a Skinny client from the Cisco CallManager. The UnregisterClient script function parameters are described in Table 3-36.
Note: Messages marked with an asterisk (*) are supported by the Cisco IP phone 7960 or 7940 style devices and are not used by 12- and 30- set emulations.
Note: If a message is not applicable for the device, it is skipped together with the pertaining response, if one is expected.
Output Settings
Ok The client registration was done successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an error.
The default resolution for this output is FAILED.
Note: Fields highlighted in blue (as shown below) in the script function configuration tabs indicate that expressions using scenario variables and numerical values are accepted as input in these fields.
Table 3-35. RegisterClient Properties (Continued)
Properties Description
Table 3-36. UnregisterClient Properties
Properties Description
Parameters Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
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OffHook Notifies the Cisco CallManager that the station is in an off-hook state. The Off-Hook script function parameters are described in Table 3-37.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout dura-tion, which can be a Global constant, a value in ms, or a timeout from the VoIP Constants pool.
• Random Between Expressions – A ran-dom timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok The client deregistration completed successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an error.
The default resolution for this output is FAILED.
Table 3-36. UnregisterClient Properties (Continued)
Properties Description
Table 3-37. OffHook Properties
Properties Description
Parameters OffHook HCall A new automatically-generated, unique call reference with a CallHandle#%d format. For every Skinny function that creates a new call reference (HCall), such as OffHook, the call handle number is increased.
For example, considering that the first OffHook function that has been added to the scenario has an HCall equal to CallHandle#1, the second OffHook function, or any other function that creates a new call reference, added to the scenario, has a CallHandle#2 call reference.
Note: Call handles cannot be changed, they are read-only.
Line No The station line number
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Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants category.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Con-stants, such as for example Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Note: If the function is made up of more than one Skinny message, then the timeout is set for the whole duration of the command.
Output Settings Ok The notification was sent successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-37. OffHook Properties (Continued)
Properties Description
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OnHook Notifies the Cisco CallManager that the station is in an on-hook state and discon-nects all active calls. The OnHook script function parameters are described in Table 3-38.
Table 3-38. OnHook Properties
Properties Description
Parameters Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
• Static Expression – A fixed timeout duration, which can be Global constant, a value in ms, or a timeout from the VoIP Constants pool.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as for example Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok The notification was sent successfully.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while sending the notification.
The default resolution for this output is WARNING.
Error The function has returned an error.
The default resolution for this output is FAILED.
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NewCall Sends a SoftKeyEvent message to the Cisco CallManager, requesting a dial tone. The NewCall script function parameters are described in Table 3-39.
Table 3-39. NewCall Properties
Properties Description
Parameters NewCall HCall
A new automatically-generated, unique call reference with a CallHandle#%d format
Line No The station line number
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants category.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok A new call was successfully created.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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EndCall Sends a SoftKeyEvent message to the Cisco CallManager, requesting a specific call completion. The EndCall script function parameters are described in Table 3-40.
Table 3-40. EndCall Properties
Properties Description
Parameters Call Handle A reference to an established call from those that have been created in the scenario by Skinny functions such as OffHook or NewCall.
If the AUTO option is selected, the call handle corresponds to the active call.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
Wait Other Party To Disconnect
If selected, the station waits for a Callstate message with a TsOnHook value. If it does not receive the message within an user-defined timeout, it sends an Endcall. The timeout value can be specified as either of the following:
• Static Expression: The timeout is specified by a static expression.
• Random between (from [value] to [value]): The timeout is a random value within a spec-ified interval.
• Wait for other party: The timeout value is a constant from the VoIP Plug-in Resource Pool |Constants category.
Output Settings
Ok The call ended successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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MakeCall Originates a call by dialing the phone number and performing the call establish-ment. The MakeCall script function parameters are described in Table 3-41.
Table 3-41. MakeCall Properties
Properties Description
Parameters NewCall HCall A new automatically-generated, unique call reference with a CallHandle#%d format
Line The station line number
Destination Specifies the call destination phone number as either of the following:
• Use Dial Plan Settings: The call destination is given by the Dial Plan page settings.
• Phone: The call destination is given by a user-defined string representing a valid phone number.
• Last Parked Call: The call destination is the last parked call.
• Last Dialed Number: Instructs the CallMan-ager to use the last dialed number.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout dura-tion, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants category.
• Random Between Expressions – A ran-dom timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME
The default is Static Expression, 20000 ms.
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WaitCall Waits for an incoming call. The WaitCall script function parameters are described in Table 3-42.
Output Settings
Ok A call was successfully established.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while sending the notification.
The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-41. MakeCall Properties (Continued)
Properties Description
Table 3-42. WaitCall Properties
Properties Description
Parameters HCall A new automatically-generated, unique call reference with a CallHandle#%d format.
Line No The line number on which the call is to be established. If the ANY option is selected, the station listens on all available lines.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
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AnswerCall This function, which is used conjointly with the previous WaitCall function, answers an incoming call by going off-hook and performing the call establish-ment. The AnswerCall script function parameters are described in Table 3-43 on page 3-54.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants category.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok A call was successfully established.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while sending the notification.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-42. WaitCall Properties (Continued)
Properties Description
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Table 3-43. AnswerCall Properties
Properties Description
Parameters Call Handle A reference to an established call. If the AUTO option is selected, the call handle corresponds to the active call.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout dura-tion, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants category.
• Random Between Expressions – A ran-dom timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok A call was successfully answered.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while sending the notification.
The default resolution for this output is WARNING.
Disconnected On the established connection, the other party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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DialDigits Dials the specified DTMF digits. The DialDigits script function parameters are described in Table 3-44.
Table 3-44. DialDigits Properties
Properties Description
Parameters Dial String Specifies the dialed digits as either of the following:
• Use Dial Plan Settings: The dial string is the destination specified in the Dial Plan page.
• Phone: A user-defined string representing a valid phone.
• Last Parked Call: The dial string corre-sponds to the last parked call.
Delay Between Digits
The inter-digit delay, which can be specified as:
• Static Expression – A fixed delay specified as a value in milliseconds (ms) or as a delay constant from the VoIP Plug-in Resource Pool | Constants category.
• Random Between Expressions – A ran-dom delay duration value in the specified interval.
• A user-defined delay specified as a constant from the VoIP Plug-In Resource Pool | Con-stants.
The default is Static Expression, 100 ms.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
Output Settings
Ok Digits were successfully dialed.
The default resolution for this output is SUCCESS.
Disconnected On the established connection, the other party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error.
The default resolution for this output is FAILED.
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WaitDigits Waits for a StationKeypadMessage sent by the Cisco CallManager to the station and detects a DTMF digits sequence. The WaitDigits script function parameters are described in Table 3-45.
Table 3-45. WaitDigits Properties
Properties Description
Parameters DTMF Detection Settings
Specifies the detection mode settings, as follows:
• Detect continuously for – Detects all digits arrived within the specified period of time (Default = 1000 s).
• Detect exactly <x> DTMFs – Detects the specified number of digits (Default = 6).
• Use Talk Time – Detects DTMFs for the duration of the TalkTime test configuration parameter.
• Detect DTMF sequence – Detects an expected sequence, user-defined or speci-fied by selecting a DTMF Sequence Pool entry.
Terminate conditions
Maximum Delay between DTMFs: The maximum amount of time, in milliseconds (ms), allowed between consecutive digits for a proper detection. After this period elapses, the function terminates. The range of values is 0 to 999999 ms. The default value is 500 ms.
First DTMF Timeout: The time, in milliseconds (ms), allowed for receiving the first digit. After this period elapses, the function exits on the Timeout output. The range of values is 0 to 999999 ms. The default value is 400 ms.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants category.
The default is Static Expression, 0 ms.
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HoldCall Performs a Hold operation on a specified call reference. The HoldCall script function parameters are described in Table 3-46.
Output Settings
Ok The digits were successfully detected.
The default resolution for this output is SUCCESS.
Timeout The digits were not detected due a timeout.
The default resolution for this output is WARNING.
Disconnected On the established connection, the other party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error.
The default resolution for this output is FAILED.
Table 3-45. WaitDigits Properties (Continued)
Properties Description
Table 3-46. HoldCall Properties
Properties Description
Parameters Call Handle A reference to the call to be put on hold. If the ANY option is selected, the currently active call is put on hold.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
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RetrieveCall Performs a Retrieve operation on a specified call reference. The RetrieveCall script function parameters are described in Table 3-47.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok The call was successfully put on hold.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-46. HoldCall Properties (Continued)
Properties Description
Table 3-47. RetrieveCall Properties
Properties Description
Parameters Call Handle A reference to the call to be retrieved from the hold state. If the AUTO option is selected, the call handle corresponds to the last active call.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
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Setup XFER Initiates a Transfer or a Conference procedure, without effectively completing the procedure. The actual transfer is performed using the complementary Complete XFER script function. The Setup XFER script function parameters are described in Table 3-48.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Constants pool.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok The call was successfully retrieved from the hold state.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-47. RetrieveCall Properties (Continued)
Properties Description
Table 3-48. Setup XFER Properties
Properties Description
Parameters NewCall Hcall A new automatically-generated, unique call reference with a CallHandle#%d format
XFer Option TRANSFER CALL or CONFERENCE CALL.
XFer CallHandle
A reference to the call on which to perform the action
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Destination Defines the call transfer destination as either of the following:
• Use Dial Plan Settings: The transfer destina-tion is given by the Dial Plan page settings.
• Phone: A user-defined string representing a valid phone number.
• Last Parked Call: The transfer destination is the last parked call.
• Last Dialed Number: Instructs the CallMan-ager to use the last dialed number.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Table 3-48. Setup XFER Properties (Continued)
Properties Description
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Complete XFER Completes a Transfer or Conference procedure that was previously initiated using the Setup XFER function. The CompleteXFER script function parameters are described in Table 3-49 on page 3-62.
Output Settings
Ok The transfer/conference procedure was successfully initiated, that is, a call was established with the third party.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while sending the notification.
The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-48. Setup XFER Properties (Continued)
Properties Description
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Table 3-49. Complete XFER Properties
Properties Description
Parameters Call Handle A reference to the call on which to perform the action
XFer Option TRANSFER CALL or CONFERENCE CALL
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
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Transfer This function, which is a combination of the Setup XFER and the Complete XFER functions, transfer the call to another party. The Transfer script function parameters are described in Table 3-50 on page 3-64.
Output Settings
Ok The transfer/conference procedure was successfully completed. For a transfer, the initial call is closed (for transfer) and the other parties should be connected. For a conference procedure, the call is connected into a conference.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while sending the notification.
The default resolution for this output is WARNING.
Disconnec-ted
On the established connection, the remote party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-49. Complete XFER Properties (Continued)
Properties Description
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Table 3-50. Transfer Properties
Properties Description
Parameters NewCall Hcall A new automatically-generated, unique call reference with a CallHandle#%d format
Destination The destination to which the call is transferred, which can be either of the following:
• Use Dial Plan Settings: The transfer desti-nation is given by the Dial Plan page set-tings.
• Phone: A user-defined string representing a valid phone number
• Last Parked Call: The transfer destination is the last parked call.
• Last Dialed Number: Instructs the CallMan-ager to use the last dialed number.
Transfer Mode Defines the transfer mode, with or without consultation of the party to which the call is transferred:
• Blind Transfer: A new call is initiated with-out consulting the party to which the call is transferred. The call is established without waiting for the third party to answer, and the callee should not answer before transfer is completed. If the third party is an IxLoad emulated Skinny phone executing an AnswerCall script function, the use of a delay before the function execution is recommended.
• Consultative transfer with duration: Before transferring the call, the remote party is being consulted for a specified or random period of time.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
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Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout dura-tion, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants.
• Random Between Expressions – A ran-dom timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok The transfer procedure was successfully performed.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while establishing the transfer.
The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-50. Transfer Properties (Continued)
Properties Description
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ForwardAllCalls Sends a SoftKeyEvent message to the Cisco CallManager, requesting that all incoming calls be forwarded. The ForwardAllCalls script function parameters are described in Table 3-51.
Table 3-51. ForwardAllCalls Properties
Properties Description
Parameters Destination The destination to which calls are forwarded, which can be specified as either of the following:
• Use Dial Plan Settings: The destination is chosen based on dial plan settings.
• Phone: The destination is specified by the specified phone number.
• Last Dialed Number: Instructs the CallMan-ager to use the last dialed number.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Constants pool.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok The forward configuration was successfully performed.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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ParkCall Parks a specified call. The ParkCall script function parameters are described in Table 3-52.
Note: Since during the first execution, the ForwardAllCalls script function configures the forwarding functionality on the CCM, it needs to be executed twice to effectively forwards the call.
Table 3-52. ParkCall Properties
Properties Description
Parameters Call Handle A reference to an established call. If the ANY option is selected, the active call is parked.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout dura-tion, which can be a Global constant, a value in ms, or a timeout from the VoIP Constants pool.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
Ok The call was successfully parked.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Note: The call parking and call pick up operations need to be executed on the same test scenario channel.
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GetCallInfo Retrieves the call information (CallInfo and CallState) into the Skinny variables supported by the IxLoad VoIP Plug-In. The GetCallInfo script function parame-ters are described in Table 3-53.
MeetMe Sets up a MeetMe conference with the test script function parameters described in Table 3-54 on page 3-69. To set up the conference, a Meet-Me softkey mes-sage is sent to the Cisco CallManager and then the Meet-Me conference call number is dialed.
Stations that want to connect to the conference have to dial in to the conference using the MakeCall script function.
Table 3-53. GetCallInfo Properties
Properties Description
Parameters Call Handle A reference to an established call
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Output Settings
Ok The retrieval of call parameters into Skinny variables was completed successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Note: For a list of the supported VoIP Skinny variables, refer to the Skinny Library Predefined Variables on page 4-8.
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Table 3-54. MeetMe Properties
Properties Description
Parameters NewCall Hcall A new automatically-generated, unique call reference with a CallHandle#%d format
Line No The station line number
Destination The call destination, which can be either of the following:
• Use Dial Plan Settings: The destination is given by the Dial Plan page settings.
• Phone: A user-defined string representing a valid phone number
• Last Parked Call: The destination is the last parked call.
• Last Dialed Number: Instructs the CallMan-ager to use the last dialed number.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
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RemoveLastConferenceParty
Sends a RmLstC softkey to the Cisco CallManager that removes from a confer-ence the party that has connected last. The RemoveLastConferenceParty script function parameters are described in Table 3-55.
Output Settings
Ok The MeetMe-type conference call setup completed successfully.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while establishing the conference call.
The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-54. MeetMe Properties (Continued)
Properties Description
Table 3-55. RemoveLastConferenceParty Properties
Properties Description
Parameters Call Handle A reference to an established call. If the AUTO option is selected, the call that last joined the conference is removed.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Output Settings
Ok The RmLstC softkey was sent successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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SendStimulus A Skinny station uses this message to inform the Cisco CallManager that a func-tional stimulus was pressed. The SendStimulus script function parameters are described in Table 3-56.
Table 3-56. SendStimulus Properties
Properties Description
Parameters Line No The station line on which the stimulus is transmitted.
Device Stimulus
The stimulus that is transmitted to the CallManager, which can be either of the following:
• SsLastNumberRedial = 1,• SsSpeedDial = 2,• SsHold = 3,• SsTransfer = 4,• SsForwardAll = 5,• SsForwardBusy = 6,• SsForwardNoAnswer = 7,• SsDisplay = 8,• SsLine = 9,• SsT120Chat = 0xA,• SsT120Whiteboard = 0xB,• SsT120ApplicationSharing = 0xC,• SsT120FileTransfer = 0xD,• SsVideo = 0xE,• SsVoiceMail = 0xF,• SsAnswerRelease = 0x10,• SsAutoAnswer = 0x11,• SsSelect = 0x12,• SsPrivacy = 0x13,• SsServiceURL = 0x14,• SsMaliciousCall = 0x1B,• SsGenericAppB1 = 0x21,• SsGenericAppB2 = 0x22,• SsGenericAppB3 = 0x23,• SsGenericAppB4 = 0x24,• SsGenericAppB5 = 0x25• SsMeetMeConference=0x7b,• SsConference=0x7d,• SsCallPark=0x7e• SsCallPickup=7f• SsGroupCallPickup=80
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
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SendSoftkey CP-7940/7960 stations use this message to inform the Cisco CallManager of a softkey event. The SendSoftKey script function parameters are described in Table 3-57.
Output Settings
<stimulus name>
This output, named after the stimulus selected in the Parameters page, indicates that the stimulus was successfully sent.
For custom stimuli, the value must have the ‘User defined #%d’ format.
Error The function has returned an error following an inexistent line number or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-56. SendStimulus Properties (Continued)
Properties Description
Table 3-57. SendSoftkey Properties
Properties Description
Parameters Call Handle A reference to an established call
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
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IsSoftKeyAvailable Verifies if a specified soft key is available on the Skinny station. The IsSoftKeyAvailable script function parameters are described in Table 3-58.
SoftKey • SkCFwdBusy
• SkCFwdNoAnswer
• SkBackSpace
• SkEndCall
• SkResume
• SkAnswer
• SkInfo
• SkConfrn
• SkPark
• SkJoin
• SkMeetMeConfrn
• SkCallPickUp
• SkGrpCallPickUp
• SkRmLstC
• SkSelect
• SkDirTrFt
• SkCongList
• SkRedial
• SkNewCall
• SkHold
• SkTrnsfer
• SkCFwdAl
Output Settings
<softkey name>
The output, named after the softkey selected in the Parameters page, indicates that the softkey was successfully sent.
Error The function has returned an error following an inexistent call handle or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Note: This function is available only for devices that have completed registration as devices compatible with CP-7940/60.
Table 3-57. SendSoftkey Properties (Continued)
Properties Description
Table 3-58. ISoftKeyAvailable Properties
Properties Description
Parameters Softkey The softkey value that is verified for availability
Use Specifies the active line or the call subject to verification.
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WaitForEvent The Skinny Client searches the message queue for a specified message in a given scope and waits for it if the message is not present. The WaitForEvent script function parameters are described in Table 3-59.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Output Settings
<softkey name>
This output, named after the softkey selected in the Parameters page, indicates that the softkey is available.
Error The softkey selected in the Parameters page is not available.
The default resolution for this output is FAILED.
Table 3-58. ISoftKeyAvailable Properties (Continued)
Properties Description
Table 3-59. WaitForEvent Properties
Properties Description
Parameters Scope Specifies the search domain as one of the following:
• Call Handle: The message is waited for in the specified call.
• Call State: The message is searched for in the message queue beginning with the specified state.
Generate New HCall
When selected, the receiving of the expected message generates a new call if the call ID is not associated with an existing call handle.
Skinny Message
The Skinny message for which to wait. For messages that configure parameters with multiple values, the drop-down list below permits you to select a value.
For example, when selecting the SendDtmfToneMessage in the Skinny Message drop-down list, the awaited value can be any DTMF or a specific DTMF, which can be selected from the drop-down list below.
• StationActivateCallPlaneMessage
• StationBackSpaceReqMessage
• StationTemplateMessage
• StationCallInfoMessage
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• StationCallStateMessage
• ANY
• TsIdle
• TsOffHook
• TsOnHook
• TsRingOut
• TsRingIn
• TsConnected
• TsBusy
• TsCongestion
• TsHold
• TsCallWaiting
• TsCallTransfer
• TsCallPark
• TsProceed
• TsCallRemoteMultiline
• TsInvalidNumber
• StationCapabilitiesReqMessage
• StationClearDisplay
• StationClearNotifyMessage
• StationClearPromptStatusMessage
• StationCloseReceiveChannel
• StationConfigStatMessage
• StationConnectionStatisticsReq
• StationDeactivateCallPlaneMessage
• StationDefineTimeDate
• StationDisplayNotifyMessage
• StationDisplayPromptStatusMessage
• StationDisplayTextMessage
• StationForwardStatMessage
• StationKeepAliveAckMessage
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• StationLineStatMessage
• StationOpenReceiveChannel
• StationRegisterAckMessage
• StationRegisterRejectMessage
• StationRegisterTokenAck
• StationRegisterTokenReject
• StationReset
• StationSelectSoftKeysMessage
• StationServerResMessage
• StationSetLampMessage
• ANY
• StationLampOff
• StationLampOn
• StationLampWink
• StationLampFlash
• StationLampBlink
• StationSetMicro-ModeMessage
• ANY
• StationMicOn
• StationMicOff
• StationSetRingerMessage
• ANY
• StationRingOff
• StationInsideRing
• StationOutsideRing
• StationFeatureRing
• StationPrecedenceRing
• StationSetSpeakerModeMessage
• ANY
• StationSpeakerOn
• StationSpeakerOff
• StationSoftKeySetResMessage
• StationSoftKeyTemplateResMessage
• StationSpeedDialStatMessage
• StationStartMediaTransmission
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• StationStartMulticastMediaReception
• StationStartMulticastMediaTransmission
• StationStartSessionTransmission
• StationStartToneMessage
• ANY
• DtSilence
• DtDtmf1
• DtDtmf2
• DtDtmf3
• DtDtmf4
• DtDtmf5
• DtDtmf6
• DtDtmf7
• DtDtmf8
• DtDtmf9
• DtDtmf0
• DtDtmfStar
• DtDtmfPound
• DtDtmfA
• DtDtmfB
• DtDtmfC
• DtDtmfD
• DtInsideDialTone
• DtOutsideDialTone
• DtLineBusyTone
• DtAlertingTone
• DtReorderTone
• DtRecorderWarningTone
• DtRecorderDetected-Tone
• DtRevertingTone
• DtReceiverOffHookTone
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• DtPartialDialTone
• DtNoSuchNumberTone
• DtBusyVerificationTone
• DtCallWaitingTone
• StationStartToneMessage
• DtConfirmationTone
• DtCampOnIndication-Tone
• DtRecallDialTone,
• DtZipZip
• DtZip
• DtBeepBonk
• DtMusicTone
• DtHoldTone
• DtTestTone
• DtPrecedenceRingBack
• DtPreemptionTone
• StationStopMediaTransmission
• StationStopMulticastMediaReception
• StationStopMulticastMediaTransmission
• StationStopSessionTransmission
• StationStopToneMessage
• StationUnregisterAckMessage
• StationVersionMessage
• StationKey-padMessage
• ANY
• skpZero
• skpOne
• skpTwo
• skpThree
• skpFour
• skpFive
• skpSix
• skpSeven
• skpEight
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• skpNine
• skpA
• skpB
• skpC
• skpD
• skpStar
• skpPound
• StationDialedNumberMessage
• StationInConference Message
• StationParkAddressMessage
• StationUserToDeviceDataMessage
• StationFeatureStatMessage
• StationDisplayPriNotifyMessage
• StationClearPriNotifyMessage
• StationStartAnnouncementMessage
• ANY
• AnnXMLCOnfigMode
• AnnOneShotMode
• AnnContinousMode
• StationStopAnnouncementMessage
• StationAnnounce-mentFinishMessage
• ANY
• playtoneOK
• playtoneEr
StationRecordInfoMessage
• StationRecorderOnhookMessage
• ANY
• rcrOK
• rcrErr
• rcrFull
• rcrMAxTime
• rcrLimit
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• StationNotifyDtmfToneMessage
• ANY
• DtDtmf1
• DtDtmf2
• DtDtmf3
• DtDtmf4
• DtDtmf5
• DtDtmf6
• DtDtmf7
• DtDtmf8
• DtDtmf9
• DtDtmf0
• DtDtmfStar
• DtDtmfPound
• DtDtmfA
• DtDtmfB
• DtDtmfC
• DtDtmfD
• StationSendDtmfToneMessage
• ANY
• DtDtmf1
• DtDtmf2
• DtDtmf3
• DtDtmf4
• DtDtmf5
• DtDtmf6
• DtDtmf7
• DtDtmf8
• DtDtmf9
• DtDtmf0
• DtDtmfStar
• DtDtmfPound
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• DtDtmfA
• DtDtmfB
• DtDtmfC
• DtDtmfD
• StationSubscribeDtmfPayloadReqMessage
• StationSubscribeDtmfPayloadResMessage
• StationSubscribeDtmfPayloadErrMessage
• StationUnSubscribeDtmfPayloadReqMessage
• StationUNSubscribeDtmfPayloadResMessage
• StationUnSubscribeDtmfPayloadErrMessage
• StationServiceUrlStatMessage
• StationSubscribeDtmf
• PayloadReqMessage
• StationCallSelectStatMachine
• StationOpenMultiMediaChannelMessage
ANY
• Media_Payload_NonStandard
• Media_Payload_G711Alaw64k
• Media_Payload_G711Alaw56k
• Media_Payload_G711Ulaw64k
• Media_Payload_G711Ulaw56k
• Media_Payload_G722_64k
• Media_Payload_G722_56k
• Media_Payload_G722_48k
• Media_Payload_G7231
• Media_Payload_G728
• Media_Payload_G729
• Media_Payload_G728AnnexA
• Media_Payload_Is11172AudioCap
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• Media_Payload_Is13818AudioCap
• Media_Payload_G729AnnexB
• Media_Payload_G729AnnexAwAnnexB
• Media_Payload_GSM_Full_Rate
• Media_Payload_GSM_Half_Rate
• Media_Payload_GSM_Enhanced_Frame
• StationOpenMulti-MediaChannelMessage
• Media_Payload_Wide_Band_256k
• Media_Payload_Data64
• Media_Payload_Data56
• Media_Payload_GSM
• Media_Payload_ActiveVoice
• Media_Payload_G726_32K
• Media_Payload_G726_24K
• Media_Payload_G726_16K
• Media_Payload_H261
• Media_Payload_H263
• Media_Payload_T120
• Media_Payload_H224
• StationStartMultiMediaTransmission
• ANY
• Media_Payload_NonStandard
• Media_Payload_G711Alaw64k
• Media_Payload_G711Alaw56k
• Media_Payload_G711Ulaw64k
• Media_Payload_G722_64k
• Media_Payload_G722_56k
• Media_Payload_G722_48k
• Media_Payload_G7231
• Media_Payload_G728
• Media_Payload_G729
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• Media_Payload_G728AnnexA
• Media_Payload_Is11172AudioCap
• Media_Payload_Is13818AudioCap
• Media_Payload_G729AnnexB
• Media_Payload_G729AnnexAwAnnexB
• Media_Payload_GSM_Full_Rate
• Media_Payload_GSM_Half_Rate
• Media_Payload_GSM_Enhanced_Frame
• Media_Payload_Wide_Band_256k
• Media_Payload_Data64
• Media_Payload_Data56
• Media_Payload_GSM
• Media_Payload_ActiveVoice
• Media_Payload_G726_32K
• Media_Payload_G726_24K
• Media_Payload_G726_16K
• Media_Payload_H261
• Media_Payload_H263
• Media_Payload_T120
• Media_Payload_H224
• StationStopMultiMediaTransmission
• StationMiscella-neousCom-mandMessage
• ANY
• videoFreezePiccture
• videoFastUpdatePicture
• videoFastUpdateGOB
• videoFastUpdateMB
• lostPicture
• lostPartialPicture
• RecoveryReferencePicture
• TemporalSpatialTradeOff
• StationFlowControlCommandMessage
• StationCloseMultiMediaReceiveChannel
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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• StationCreateConferenceReqMessage
• RT_Conference
• RT_IVR
• StationDeleteConferenceReqMessage
• StationModifyConferenceReqMessage
• StationAddParticipantsReqMessage
• StationDropParticipantsReqMessage
• StationAuditConferenceReqMessage
• StationAuditParticipantReqMessage
Empty Message Queue
The message is deleted from the message queue.
Delay Before Execution
The time to wait before the function starts. Introducing a delay is used for synchronization reasons, such as, for example, to synchronize functions on different scenario channels.
The delay value can be specified as a static expression, a random value, or as a predefined constant from the VoIP Plug-In Resource Pool | Constants.
The default is Static Expression, 0 ms.
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. The timeout value can be specified as:
• Static Expression – A fixed timeout duration, which can be a Global constant, a value in ms, or a timeout from the VoIP Plug-in Resource Pool | Constants.
• Random Between Expressions – A random timeout duration value in a specified interval.
• Any of the timeout constants from the VoIP Plug-in Resource Pool | Constants, such as, for example, Delay between digits or PHONE_WAIT_TIME.
The default is Static Expression, 20000 ms.
Output Settings
<message to wait for>
This output, named after the message selected in the Parameters page, indicates that the message was received.
The default resolution for this output is SUCCESS.
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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Timeout A timeout occurred while receiving the message.
The default resolution for this output is WARNING.
Error The specified message has not been received.
The default resolution for this output is FAILED.
Table 3-59. WaitForEvent Properties (Continued)
Properties Description
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VoIP Media Functions Library
The VoIP Media functions are used to generate media streaming for calls estab-lished by using either VoIP protocol implementation.
The currently implemented Media functions are the following:
• Generate DTMF on page 3-86.
• Detect DTMF on page 3-88.
• Generate MF on page 3-90.
• Detect MF on page 3-92.
• Generate Tone on page 3-92.
• Wait for Tone on page 3-93.
• Talk on page 3-94.
• Listen on page 3-95.
• Voice Session on page 3-97.
• Multimedia Session on page 3-98
• T.38 Fax Session on page 3-100
• Path Confirmation on page 3-102.
• RTP Control on page 3-107
Generate DTMF Generates a specified DTMF sequence.
Generate DTMF: Parameters
Table 3-60 describes the Generate DTMF function parameters.
Table 3-60. Generate DTMF Parameters
Name Description
Delay Before Execution
The time to wait before the function starts. It can be used for synchronization reasons (that is, to synchronize the Make Call function with the Receive Call one). It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, Sleep 2000 in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
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Generate DTMF: Advanced Playback Settings
Table 3-61 describes the Generate DTMF function advanced playback settings.
The default is Static Expression, 20000 ms.
DTMF Sequence Selects a specific DTMF sequence item from those defined in the DTMF Sequence Pool Items or Local Sequence. The default is 12345.
DTMF Duration The time (in ms) required for a single tone to be generated. The range of values is 0 to 999999 ms. The default value is 100 ms.
Inter DTMF Interval The maximum amount of time (in ms) between two consecutive generated DTMFs. The range of values is 0 to 999999 ms. The default value is 200 ms.
DTMF Amplitude The attenuation (in dB) of the DTMF tone. The minimum attenuation is 0 dB (no attenuation) and the maximum is –40 dB. The default value is –10 dB.
Note: Fields highlighted blue (shown in the image below) in the script function configuration tabs indicate that expressions using scenario variables and numerical values are accepted as input in these fields.
Table 3-61. Generate DTMF Advanced Playback Settings
Name Description
Transmission Mode Generates a sequence of DTMFs. The sequence can have up to 31 tones and it can contain any combination of the standard tones, (that is, '1', '2', '3', '4', '5', '6', '7', '8', '9', '0', '#', '*', 'A', 'B', 'C', 'D'). The DTMF sequence can be transmitted In Band or Out of Band using a 2833 event payload format or the signaling layer and it can be played continuously for a number of times or for a specific period of time.
• In Band: Using RTP media streaming.
• Out of Band – Using 2833 EVENT Payload For-mat: The RTP Payload format used for carrying dual-tone multi frequency (DTMF) digits and other line and trunk signals as events.
Table 3-60. Generate DTMF Parameters (Continued)
Name Description
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Generate DTMF: Outputs
Table 3-62 describes the outputs available for the Generate DTMF function.
Detect DTMF Detects a sequence of DTMF signals.
Detect DTMF: Parameters
Table 3-63 describes the Detect DTMF function parameters.
• Out of Band – Using 2833 TONE Payload For-mat: Using 2833 TONE Payload Format –The RTP Payload format that can represent tones consisting of one or more frequencies. For details on RTP Payload, refer to RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Sig-nals.
• Use Global Settings: Use the settings from the Global Settings>Library Settings and Out-puts>VoIP>RTP page.
Playback • Play: Specifies the number of times to play.
• Repeat Continuously for: Specify the period of time to play. The default value is 1000 seconds.
• Use Talk Time (only for BHCA objective): Plays the DTMF for the duration of the Talk Time call parameter.
• Use Global Settings: Use the settings from the Global Settings>Library Settings and Out-puts>VoIP>RTP page.
Table 3-62. Generate DTMF Outputs
Output Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Disconnected On the established connection, the other party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
Table 3-61. Generate DTMF Advanced Playback Settings (Continued)
Name Description
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Detect DTMF: Advanced Detection Settings
Table 3-64 describes the Detect DTMF advanced detection settings.
Table 3-63. Detect DTMF Parameters
Name Description
Delay Before Execution
The time to wait before the function starts. It can be used for synchronization reasons (that is, to synchronize the Make Call function with the Receive Call one). It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
DTMF Detection Settings
• Detect continuously for: Detects all the digits arrived within the specified period of time. The default value is 1000 seconds.
• Detect exactly: Detects the specified number of digits. The default value is 6 DTMFs.
• Use Talk Time (only for BHCA objective): Detects DTMS for the duration of the Talk Time parameter.
• Detect DTMF Sequence: Detects the expected sequence (specified by selecting an entry in the DTMF Sequence Pool Items, or a specified Local Sequence).
Terminate Conditions • Maximum delay between DTMFs: The maximum amount of time (in ms) allowed between consecu-tive digits for a proper detection. After this period elapses, the function terminates. The range of val-ues is 0 to 999999 ms. The default value is 3000 ms.
• First DTMF Timeout: The time (in ms) allowed for receiving the first digit. After this period elapses, the function exits on Timeout output. The range of values is 0 to 999999 ms. The default value is 2000 ms.
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Detect DTMF: Outputs
Table 3-65 describes the outputs available for the Generate DTMF function.
Generate MF Generates a specified MF sequence.
Table 3-64. Detect DTMF Advanced Detection Settings
Name Description
Transmission Mode Detects DTMFs. The DTMFs can be detected In Band or Out of Band, using 2833 or the signaling layer:
• In Band – Using RTP media streaming.
• Out of Band – Using 2833 EVENT Payload For-mat: The RTP Payload format used for carrying dual-tone multi frequency (DTMF) digits and other line and trunk signals as events.
• Out of Band – Using 2833 TONE Payload For-mat: The RTP Payload format that can represent tones consisting of one or more frequencies. For details on RTP Payload, refer to RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals.
• Use Global Settings – Use the settings from the Global Settings>Library Settings and Out-puts>VoIP>RTP page.
Note:
In-band DTMF detection cannot be performed with audio codecs other than G.711 aLaw and G.711 uLaw.
Table 3-65. Detect DTMF Outputs
Output Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Timeout This output is enabled if the request is not received within the specified timeout.
The default resolution for this output is WARNING.
Disconnected On the established connection, the other party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
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Generate MF: Parameters
Table 3-66 describes the Generate MF function parameters.
Generate MF: Advanced Playback Settings
These settings are identical to the Generate DTMF Advanced Playback Settings. For more information, please refer to Generate DTMF: Advanced Playback Settings on page 3-87.
Generate MF: Outputs
These outputs are identical to the Generate DTMF Outputs. For more informa-tion, please refer to Generate DTMF: Outputs on page 3-88.
Table 3-66. Generate MF Parameters
Name Description
Delay Before Execution
The time to wait before the function starts. It can be used for synchronization reasons (that is, to synchronize the Make Call function with the Receive Call one). It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
MF Settings • MF Duration: The time (in ms) required by a single tone to be generated. The range of values is 0 to 999999 ms. The default value is 200 ms.
• Inter MF Interval: The maximum amount of time (in ms) between two consecutive generated MFs. The range of values is 0 to 999999 ms. The default value is 200 ms.
• MF Amplitude: The attenuation (in dB) of the MF tone. The minimum attenuation is 0 dB (no attenua-tion) and the maximum is –40dB. The default value is –10 dB.
MF Sequence Selects a specific MF sequence item from those defined in the DTMF Sequence Pool Items, or a Local Sequence. The default is 12345.
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Detect MF Detects a sequence of MF signals.
All Detect MF function parameters are identical to these of the Detect DTMF function. For more information, please refer to Detect DTMF on page 3-88.
Generate Tone Generates a custom tone.
Generate Tone: Parameters
Table 3-67 describes the Generate Tone function parameters.
Note:
In-band MF detection cannot be performed with audio codecs other than G.711 aLaw and G.711 uLaw.
Table 3-67. Generate Tone Parameters
Name Description
Delay Before Execution
The time to wait before the function starts. It can be used for synchronization reasons (that is, to synchronize the Make Call function with the Receive Call one). It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
Tone Settings • Tone Duration: The time (in ms) required to gener-ate a single tone. The range of values is 0 to 999999 ms. The default value is 100 ms.
• Tone Amplitude: The attenuation (in dB) of the DTMF tone. The minimum attenuation is 0 dB (no attenuation) and the maximum is –40dB. The default value is –10 dB.
Custom Tone • Tone Name: Selects a specific tone from the drop-down list.
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Generate Tone: Advanced Playback Settings
All parameters in the Generate Tone Advanced Playback Settings tab are identi-cal to the Generate DTMF Advanced Playback parameters. For more informa-tion, please refer to Generate DTMF: Advanced Playback Settings on page 3-87.
Generate Tone: Outputs
These outputs are identical to the Generate DTMF Outputs. For further informa-tion, please refer to Generate DTMF: Outputs on page 3-88.
Wait for Tone Detects a custom tone.
Wait for Tone: Parameters
Table 3-68 describes the Wait for Tone function parameters.
Wait for Tone: Advanced Detection Settings
All parameters in the Wait for Tone Advanced Detection Settings tab are identi-cal to the Detect DTMF Advanced Detection parameters. For more information, please refer to Detect DTMF: Advanced Detection Settings on page 3-89.
Table 3-68. Wait for Tone Parameters
Name Description
Delay Before Execution
The time to wait before the function starts. It can be used for synchronization reasons (that is, to synchronize the Make Call function with the Receive Call one). It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
Tone Detection Settings
Select/Add the tone(s) to detect from those available in the Custom Tones Pool.
Terminate Conditions Tone Timeout: The time (in ms) allowed for receiving a tone. After this period elapses, the function exits on Timeout output. The range of values is 0 to 999999 ms. The default value is 500 ms.
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Wait for Tone: Outputs
Table 3-69 on page 3-94 describes the outputs available for the Wait For Tone function.
Talk Plays the specified wave files across the established call.
Talk: Parameters
Table 3-70 describes the Talk function parameters.
Note:
In-band tone detection cannot be performed with audio codecs other than G.711 aLaw and G.711 uLaw.
Table 3-69. Wait for Tone Outputs
Output Name
Description
Timeout The timeout value was exceeded or the received sequence does not correspond with the expected one.
The default resolution for this output is WARNING.
Disconnected On the established connection, the other party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
Table 3-70. Talk Parameters
Name Description
Delay before execution
The time to wait before the function starts. It can be used for synchronization reasons (that is, to synchronize the Make Call function with the Receive Call one). It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
Overwrite activity settings
If selected, the selection of the audio clip and the playback settings specified at activity level are overrriden by the current settings.
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Talk: Advanced Playback Settings
Table 3-71 describes the Talk function advance playback parameters.
Talk: Outputs
These outputs are identical to the Generate DTMF Outputs. For further informa-tion, please refer to Generate DTMF: Outputs on page 3-88.
Listen Listens to an audio RTP stream for the specified duration.
Clip An audio clip from the Resource Pool – either a simple .wav files or an speech clip – to be played back by the
script function. Clicking the : opens a window that permits you to select a .wav file.
Note: The eduration of speech clips, which are used for P.862 PESQ and P56 QoV computation, must not exceed 30 seconds.
Output level The clip output level (default -20 dbm).
Playback Settings - This area permits you to choose a clip playback duration.
Play for clip duration If selected, the clip is played entirely for a specified number of times.
Repeat continuously for
If selected, the clip is played repeatedly for a specified duration of time.
Use Talk Time (for all objectives except channels)
If selected, the clip is played for the duration of the Talk Time parameter, an important parameter configured in the case of BHCA/CPS/LPS test objectives.
Use Global settings If selected, the clip is played as specified in the RTP page of the Global Settings window.
Table 3-71. Talk Advanced Playback Parameters
Name Description
Terminate conditions Specifies a condition for stopping the wave playback as either of the following:
• DTMF: If selected, playing stops when the specified DTMF digit is received. The default DTMF is 0.
• Tone: If selected, playing stops when the specified custom tone is received. You can choose one of the available custom tones in the tones pool.
• MF: If selected, playing stops when the specified MF digit is received. The default value is 0.
Table 3-70. Talk Parameters (Continued)
Name Description
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Listen: Parameters
Table 3-72 describes the Listen function parameters.
Listen: Advanced Settings
Table 3-73 describes the Listen function advanced recording parameters.
Note: Direct recording of wave files by the application is currently not supported.
Table 3-72. Listen Parameters
Name Description
Delay before execution
The time to wait before the function starts. This delay can be used for synchronization purposes, such as, for example, to synchronize the call originating side with the call terminating side. It can be specified as either of the following:
• Static Expression (ms)
• Random Between Expressions (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 1000, 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
Overwrite activity settings
If selected, the listen setting and QoV computation settings specified at activity level are overrriden by the current settings.
Listen Settings Specifies the listen duration as either of the following:
• Listen Duration: Specifies the period of time the lis-ten operation is performed. The default value is 10000 ms.
• Use Talk Time (only for BHCA objectives): Lis-tens for an audio stream for the duration of the Talk Time parameter.
Perform QoV measurements
If selected, performs P.862 PESQ and P56 QoV scores computation for the reference wave file specified in the Clip field.
Note: Speech clips duration must not exceed 30 seconds.
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Listen: Outputs
These outputs are identical to the Generate DTMF Outputs. For further informa-tion, please refer to Generate DTMF: Outputs on page 3-88.
Voice Session Plays and listens simultaneously (to) the specified wave files across the estab-lished call.
Voice Session: Talk Parameters
These parameters are identical to the Talk function parameters. For more infor-mation, please refer to Talk: Parameters on page 3-94.
Voice Session: Listen Parameters
These parameters are identical to the Listen function parameters. For more infor-mation, please refer to Listen: Parameters on page 3-96.
Voice Session: Advanced Playback Settings
Table 3-74 describes the Voice Session Advanced Playback parameters.
Table 3-73. Talk Advanced Settings
Name Description
Terminate conditions Specifies a condition for stopping the recording as either of the following:
• DTMF: If selected, playing stops when the specified DTMF digit is received. The default DTMF is 0.
• Tone: If selected, playing stops when the specified custom tone is received. You can choose one of the available custom tones in the tones pool.
• MF: If selected, playing stops when the specified MF digit is received. The default value is 0.
Note: The audio files played by the Voice Session script function can be either simple wave files or audio clips (.wav) with a duration of maximum 30 seconds. Additionally, the files need to have the following characteristics:
• Coding: PCM, A-LAW. MU-LAW
• Sampling frequency: 8 kHz, monophonic, and having 8 bit/sample.
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Voice Session: Output Settings
Table 3-75 describes the outputs available for the Voice Session function.
Multimedia Session Plays audio and video files simultaneously or independently, depending on the current script function configuration, across an established SIP or H323 call.
Multimedia Session: Video Play Parameters
Table 3-76 describes the Multimedia Session Video Play parameters.
Table 3-74. Voice Session Advanced Playback Settings
Name Description
Terminate Conditions - This area specifies conditions for terminating the playback and the recording of audio clips.
Stop playback on first detected
If selected, this option enables you to specify a condition for stopping the audio playback as either of the following:
• DTMF: If selected, playing stops when the specified DTMF digit is received. The default DTMF is 0.
• Tone: If selected, playing stops when the specified custom tone is received. You can choose one of the available custom tones in the tones pool.
• MF: If selected, playing stops when the specified MF digit is received. The default value is 0.
Stop recording on first detected
If selected, this option enables you to specify a condition for stopping the audio recording as either of the following:
• DTMF: If selected, playing stops when the specified DTMF digit is received. The default DTMF is 0.
• Tone: If selected, playing stops when the specified custom tone is received. You can choose one of the available custom tones in the tones pool.
• MF: If selected, playing stops when the specified MF digit is received. The default value is 0.
Table 3-75. Voice Session Outputs
Output Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
Note: The video files played by the Multimedia Session function need to have an MP4 format and be no larger than 80 MB.
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Multimedia Session: Advanced Settings
Table 3-77 describes the Multimedia Session Advanced parameters.
Table 3-76. Multimedia Session Video Play Settings
Name Description
Delay before execution
The time to wait before the function starts. It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
Overwrite playback activity settings
For ease of configuration, playback settings for voice and video media can normally be configured at activity level. When this option is selected, the activity-level settings for the script function can be overriden.
Play audio When the Overwrite playback activity settings option is selected, also selecting this option enables you to define an audio clip to play.
Play video When the Overwrite playback activity settings option is selected, also selecting this option enables you to define an mp4 video clip (AVC-format) to play.
Note: In the case of VoIPSIP tests, you can also select SVC-format h264 video files to be played by this script function.
Play for clip duration or Talk Time
If selected, the clip(s) are played entirely, or for the duration of the Talk Time parameter (in case of a BHCA/CPS test objective).
Note: In the case of BHCA/CPS objectives, the Talk Time and the number of channels are the two important parameters that must be specified for a given test objective value.
Play for If selected, plays the clip(s) for a specified duration of time.
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Multimedia Session: Output Settings
Table 3-78 describes the outputs available for the Multimedia Session function.
T.38 Fax Session Sends or receives the specified fax data file across an established SIP call.
T.38 Fax Session: General Parameters
Table 3-79 describes the T.38 Fax Session function parameters.
Table 3-77. Multimedia Session Advanced Settings
Name Description
Terminate Conditions Stop playback on first detected: If enabled, playback stops if the specified DTMF, MF, or custom tone is detected (default Disabled).
DTMF: If selected, listening stops when the specified DTMF digit is received. The default DTMF is 0.
Tone: If selected, listening stops when the specified custom tone is received. You can choose one of the available custom tones in the tones pool.
MF: If selected, listening stops when the specified MF digit is received. The default is 0.
Table 3-78. Multimedia Session Output Settings
Output Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
Note: The T.38 Fax Session script function can only be used for calls negotiated using the SIP signaling protocol.
Table 3-79. T.38 Fax Session - Parameters
Name Description
Send fax If selected, the image specified in the adjoining field is sent over a simple fax session. To specify an image,
click the and choose an image file from the Resource Pool.
Receive fax If selected, the script function executes a fax reception.
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Overwrite activity settings
If selected, the activity-level fax parameters (specified in the T.38 and T.30 tabs) are overridden by function-level settings.
You can define an image to send/receive other than that configured at activity level in the T.30 tab and overrride the following send/receive parameters:
Send Parameters
Coding The highest coding scheme available for compressing the page data when sending. Possible values are MH, MR, and MMR (default).
Data rate (kbps) The data rate for sending, which is any of the following: V.27 ter 2.4 Kbps, V.27 ter 4.8 Kbps, V.17 7.2 Kbps, V.17 9.6 Kbps, V.17 12 Kbps, V.17 14.4 Kbps, V.29 7.2 Kbps, V.29 9.6 Kbps, V.34 16.8 Kbps, V.34 19.2 Kbps, V.34 21.6 Kbps, V.34 24 Kbps, V.34 26.4 Kbps, V.34 28.8 Kbps, V.34 31.2 Kbps and V.34 33.6 Kbps available data rates.
Page size The sent page size, any of the following: A4 (210x297 mm), B4 (255x364 mm), and A3 (297x420 mm) available formats.
MSLT The minimum transmission time of one coded scan line (values (lower value than specified are accepted). This value is determined automatically, based on DIS.
Protocol The protocol used for fax sending. The available protocols are non-ECM and ECM (default).
Resolution The horizontal and vertical resolution of the page image. Possible values is any combination of the following:
- R8x3.85 lines/mm (default)
- R8x7.7 lines/mm
- R8x15.4 lines/mm
- 200x200 dots/inch
Send CNG Specifies if the CNG message is sent or not (default yes).
Receive Parameters
Coding The highest coding scheme available for compressing the page data when receiving. Possible values are MH, MR, and MMR (default).
Page size (up to) The maximum size of the received page. Possible values are A4 (210x297 mm) (default), B4 (255x364 mm), and A3 (297x420 mm).
Table 3-79. T.38 Fax Session - Parameters (Continued)
Name Description
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T.38 Fax Session: Output Settings
Table 3-80 describes the T.38 Fax Session Output Settings parameters.
Path Confirmation After a voice path has been established using VoIP signaling, this script function confirms the existence of a viable voice path between two VoIP endpoints by simultaneously generating and detecting DTMF, MF or tone sequences. Path confirmation is performed in-band (using RTP audio streaming) or out-of-band (using RFC 2833 events).
MSLT The minimum transmission time of one coded scan line (values (lower than specified are accepted). The possible values are:
- 0 ms T7.7 = T3.85 (default)
Protocol The protocol used to receive fax. The available protocols are non-ECM and ECM (default).
Send CED before DIS If selected, this enables the answering fax to send a CED (CallEd station Identification) signal. This tone is generated to allow a human participant to realize that a machine is present on the other end of the call (default Disabled).
Modulation Specify the protocols available for receiving. Possible values are:
- V.27
- V.27/V.29
- V.27/V.29/V.17
- V.27/V.29/V.17/V.34 (default)
Receive Resolution - This area specifies a list of resolutions (horizontal and vertical) to choose from. Possible values are any combination of the following:
- R8x3.85 lines/mm
- R8x7.7 lines/mm
- R8x15.4 lines/mm
- 200x200 dots/inch
Table 3-80. Fax Session - Output Settings
Properties Description
Output Settings
OK The fax session was completed successfully. The default resolution for this output is SUCCESS.
Error The function execution has returned an error. The default resolution for this output is FAILED.
Table 3-79. T.38 Fax Session - Parameters (Continued)
Name Description
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Path confirmation sequences can be generated once or can be looped through for a specified period of time. The existence of the voice path is validated if all the sequences can be detected by the endpoints in the specified timeout intervals.
The Path Confirmation script function is full duplex, meaning that the generation and detection of sequences can be performed at the same time.
Path Confirmation: Parameters
Table 3-81 describes the Path Confirmation function parameters.
Notes:
• The Path Confirmation function does not support cadenced tones.
• In-band path confirmation cannot be performed with audio codecs others than G.711 aLaw and G.711 uLaw.
Note: After the Path Confirmation function starts execution and before tones generation / detection is actually performed, a synchronization procedure for functions on different scenario channels is launched, having a maximum duration of 2000 ms.
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Table 3-81. Path Confirmation - Parameters
Name Description
Delay before execution
The time to wait before the function starts. This delay can be used for synchronization purposes, such as, for example, to synchronize the call originating side with the call terminating side. It can be specified as:
• Static Expression: This fixed delay before execu-tion can be expressed as a value in milliseconds (ms), or as a formula.
• Random Between Expressions: This is a random delay in the specified interval that is used to simu-late real-life conditions. It is expressed in millisec-onds (ms).
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 1000, 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
Path Confirmation Method
Specifies the path confirmation method as using either DTMFs, MFs, or custom tones sequences.
The path confirmation initiator sends the specified digits sequence and expects to receive the same sequence, while the path confirmation receiver expects the specified digits and then sends them back.
When you choose to send sequences as DTMFs or
MFs, you need to click the corresponding to define the sequence as a VoIP Plug-in Resource Pool predefined sequence or by entering the digits directly into a field (local sequence).
When you choose to perform path confirmation using custom tones, you need to look up the predefined editable tones from the Resource Pool > Custom Tones category – such as, for example, DIGIT_1, DIGIT_2, DIGIT_3, DIGIT_4, DIGIT_5, DIGIT_6, DIGIT_7, DIGIT_8, DIGIT_9, or DIGIT_0 – and define the sequence by specifying only the corresponding digits. For example, a valid sequence comprising five tones is specified using the 01234 string.
Note:
DTMF sequences can be made up of 0-9, A, B, C, D digits, while MF sequences support only 0-9, A, B, C digits.
Note: When specifying a DTMF, MF, or tone sequence using expressions, you must ensure that every channels pair evaluates the expressions to the same value, otherwise the path confirmation sequence fails.
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Path Confirmation: Tone Detection/Generation Page
Table 3-82 on page 3-105 describes the tone detection/generation parameters.
Path Confirmation: Advanced Settings
Table 3-83 on page 3-106 describes the Path Confirmation function advanced settings.
Table 3-82. Path Confirmation - Tone Detection/Generation Parameters
Name Description
Use values from Global Settings
If selected, the values of all other parameters on this page are taken from the Global Settings>Library Settings and Outputs>VoIP>RTP page.
DTMF/MF/Tone Generation
DTMF/MF/Tone generation settings, as follows:
• Tone Duration: The duration (in ms) of a single tone (DTMF/MF/custom tone). The range of values is 40 to 59960 ms. The default value is 200 ms.
• Inter Tone Interval: The maximum amount of time (in ms) between two consecutive generated DTMFs/MFs/Custom Tones. The range of values is 40 to 59960 ms. The default value is 200 ms.
Note: The sum of the tone duration and the inter tone interval is required to be less than 60000 ms.
• Tone Amplitude: The attenuation (in dB) of the DTMF/MF/custom tone. The minimum attenuation is 0 dB (no attenuation) and the maximum is -40 dB. The default value is -10 dB.
DTMF/MF/Tone Detection
DTMF/MF/Tone detection settings, as follows:
• First Sequence Timeout: The time (in ms) allowed for detecting the first DTMF/MF/custom tones sequence. After this period elapses, the function exits on Timeout output. The range of values is 200 to 999999 ms. The default value is 5000 ms.
• Intersequence timeout: The maximum amount of time (in ms) allowed between consecutive DTMFs/MFs/custom tones sequences for a proper detec-tion. After this period elapses, the function exits on Timeout output. The range of values is 200 to 999999 ms. The default value is 5000 ms.
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Path Confirmation: Output Settings
Table 3-84 describes the Path Confirmation output parameters.
Table 3-83. Path Confirmation - Advanced Settings
Name Description
Transmission Mode Specifies the DTMF/MF/TONE transmission mode as one of the following:
• In Band: Uses RTP media streaming
• Out of Band - Using 2833 EVENT Payload Format: This is the RTP Payload format used for carrying dual-tone multi frequency (DTMF) digits and other line and trunk signals as events. See RFC2883 for details.
• Out of Band - Using 2833 TONE Payload Format: This is the RTP Payload format that can represent tones consisting of one or more frequencies. For details on RTP Payload, please refer to RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals.
• Use Global Settings: Uses the transmission mode settings from the Global Settings>Library Settings and Outputs>VoIP>RTP page.
Notes:
• A G.711 aLaw or uLaw codec must be used for both in-band and out-of-band path confirmation. If a G.711 codec cannot be found, the path confirmation opera-tion cannot start.
• A 2833 Event/Tone codec must be used for out of band Event/Tone media transmission.
Playback Specifies how many times the path confirmation sequence is executed, as either of the following options:
• Execute once: The sequence is played once.
• Execute for: The sequence is executed for a user-defined period of time, expressed in seconds, minutes or hours. The default value is 10 seconds.
Note: An expression can be also entered in this edit-box. At runtime the expression is evaluated and its value is considered in milliseconds.
• Use Global Settings: Uses the path confirmation sequence settings from the Global Set-tings>Library Settings and Outputs>VoIP>RTP page.
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RTP Control IxLoad RTP script functions support Non-Blocking Execution, meaning that, after a RTP script function has been initialized and started, the execution flow advances to the next function in scenario, without waiting for the current com-mand to finish. When running in Non-Blocking Execution mode, RTP script functions perform their tasks in background, allowing simultaneous handling of other signaling and media functions.
The RTP Control function has the purpose of controlling one (the previous) RTP script function when executing a RTP script function that has non-blocking behavior enabled.
If non-blocking execution is not enabled and the RTP Control script function is used in a test scenario, it exits on a SUCCESS output.
RTP Control: Parameters
Table 3-85 describes the RTP Control function parameters.
Table 3-84. Path Confirmation - Output Settings
Output Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Timeout This output is enabled if the request is not received within the specified timeout.
The default resolution for this output is WARNING.
Disconnected On the established connection, the other party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
Mismatch The function has detected a digit that is not in the expected path confirmation sequence.
Notes:
• An RTP script function running in non-blocking mode always exits on the OK/SUCCESS output, while the ‘true’ result of the function execution is obtained by evaluating the output of the RTP Control function. A special case is the WaitTone script function which does not have an OK output. A non-blocking WaitTone function always exits on the output of the expected first tone.
Note: Only one RTP script function can be active at any one time during test execution. If the scenario flow contains an RTP script function with non-blocking behavior and this function is followed by a second RTP function, when the execution flow reaches the second function, the first function is terminated, such that no RTP functions execute in parallel.
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RTP Control: Output Settings
Table 3-86 describes the RTP Control output parameters.
Table 3-85. RTP Control - Parameters
Name Description
Delay before execution
The time to wait before the function starts. This delay can be used for synchronization purposes, such as, for example, to synchronize the call originating side with the call terminating side. It can be specified as:
• Static Expression: This fixed delay before execu-tion can be expressed as a value in milliseconds (ms), or as a formula.
• Random Between Expressions: This is a random delay in the specified interval that is used to simu-late real-life conditions. It is expressed in millisec-onds (ms).
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 1000, 2000, in milliseconds (ms)
The default is Static Expression, 20000 ms.
Control Actions • Check for RTP Completion: Checks the status of an RTP function executing in non-blocking mode, and returns the status code (see RTP Control: Output Settings) accordingly.
• Terminate RTP: Forcefully terminates a running non-blocking RTP function.
Table 3-86. RTP Control - Output Settings
Output Name Description
RTP Running This output is selected when a non-blocking RTP script function is running in background (cannot be returned if the Terminate RTP option was configured for the RTP Control function).
The default resolution for this output is SUCCESS.
RTP Not Running The previous RTP function is no longer running in the background and terminated successfully.
No RTP function was started before.
The default resolution for this output is SUCCESS.
Error The RTP function running in the background exited with an error.
The default resolution for this output is FAILED.
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Forceful RTP Function Termination and Statistics
An RTP function that is running in the background is forcefully terminated in any of the cases below:
• When SDP renegotiation changes the RTP parameters (codec, etc.): In this case, the RTP Skipped Functions statistic is incremented.
• When the RTP Control function terminates it: In this case, the RTP Skipped Functions statistic is incremented.
• When a subsequent RTP script function terminates it: In this case, the RTP Skipped Functions statistic is incremented.
• When an EndCall function is executed while the RTP function is running and the EndCall procedure must be initiated: In this case, the RTP Disconnected Functions statistic is incremented.
• When Graceful ramp-down is started: In this case, an RTP function is stopped/skipped and the RTP Skipped Functions statistic is incremented.
• When normal ramp-down is started: In this case, no statistics are incre-mented.
• When the call ends and the EndCall procedure is initiated: In this case, the RTP Errors and the Failed Playbacks/Failed Records statistics are incre-mented.
Flow Statistics
Flow statistics are updated normally, since a non-blocking RTP function always exits on the SUCCESS output and the real result of the function execution is obtained from the RTP Control function.
SDP Renegotiation
If SDP re-negotiation is done after an RTP non-blocking function has started, the function is stopped and RTP context is re-opened with new parameters. The stopped RTP function is not restarted after the RTP context is re-opened.
Timeout The RTP function running in the background has finished with a timeout condition.
The default resolution for this output is WARNING.
Disconnected The RTP function running in the background has finished with a disconnect condition.
The default resolution for this output is WARNING.
NOTE: The Wait Tone script function is an exception and in case the output for a Wait Tone is a detected tone (no timeout or error), a variable called RTP_NB_DETECTED_TONE is filled with the name of the detected tone and RTP Control function returns RTP Not Running.
Table 3-86. RTP Control - Output Settings (Continued)
Output Name Description
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If after a SDP renegotiation nothing has changed, then the existing RTP context is unchanged, and any running RTP non-blocking function is not stopped.
Warnings and Errors
• If non-blocking execution is enabled, but the test does not contain a RTP Control script function, the following warnings are shown when applying the test configuration:
RTP Non-blocking execution is enabled, but the test does not contain a RTP Control script function. The status of the RTP script functions execu-tion will not be available.
• If non-blocking execution is disabled but the scenario does contain a RTP Control script function the following warning are shown when applying the test configuration:
RTP Non-blocking execution is disabled, but the test contains a RTP Con-trol script function. RTP script functions execution always exit on RTP Not Running output.
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VoIP Flow Functions Library
The VoIP Flow Test Library includes the following script functions:
• Start on page 3-111.
• Stop on page 3-111.
• Variable Set on page 3-111.
• Variable Test on page 3-112.
• Sleep on page 3-113.
• Procedure on page 3-114.
• Exit Procedure on page 3-114.
• Counter Op on page 3-114.
• Test Time on page 3-115.
• Log Message on page 3-115.
• Dump Variables on page 3-115.
• Error Handler on page 3-116.
Start Description
Indicates the beginning of a test scenario flow on the channel.
Parameters
No parameters
Stop Description
Indicates the end of an execution thread or the end of the entire test scenario.
Parameters
No parameters
Variable Set Description
Declares and sets the variable values to use in the test scenario.
For each variable, you must specify a name, a type (long, string, array of long, or array of string) and a value, or an expression composed of numbers and variables.
NOTE: Only one Start function is allowed per channel.
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When specifying an expression instead of a value, at execution time, the expres-sion is evaluated to a value that is assigned to the variable.
Parameters
The user-defined variables used in this function block, of whom only the selected (tagged) variables are considered. The tagged variables display first.
Variables can be assigned numeral values, or expressions comprising numerals, operators, functions and/or variables.
A couple of expression examples are given in Table 3-87.
Variable Test Description
This function assesses a series of expression sequences, each of whom adds a new output to the script function. At the time of the script function execution, the expression sequences, composed of numerals, operators, functions, and/or vari-ables, are evaluated to either True or False.
The evaluation starts with the first expression sequence and continues until a sequence is found that evaluates to True. When such an expression sequence is found, the execution flow goes on with the block connected to the output related to that sequence.
If none of the assessed sequences return true, the execution flow goes on with the block connected to the Mismatch output.
Note: The Variable Set function is generally used conjointly with the Variable Test function that enables you to evaluate custom expressions.
Note: Evaluating an expression assigned to a variable is done by the Expression Evaluator, whose supported operators and operations are described in Appendix B, The Expression Evaluator Syntax.
Table 3-87. Expression Examples
Expression Comments
$MapPos + 2 Used in a Variable Set function to increment the value of the $MapPos variable ($MapPos is equal to the current mapping position number).
$NewMediaPort = $VoipMediaBaseAddress + $UnitCh
When used in a custom SDP message, this expression assigns to the $NewMediaPort variable the value of the base RTP port (usually 10000) increased by value of the channel index. For example, unit channel 0 will have a media port equal value of 10000, while unit channel 10 will have the RTP port value of 10010.
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Parameters
The associated parameters are the series of expressions evaluated in the function at test execution time.
Each expression has a name that generates an associated output, and an expres-sion sequence. While evaluating the expressions in turn, if the script function encounters an expression that evaluates to True, the Variable Test function exits on the corresponding output.
Operations available for creating and managing expressions are given in Table 3-88.
Sleep Description
This script function inserts a static or random delay to the execution flow.
Parameters
The associated parameter is the delay applied to the execution flow. The delay may be:
• Static – the fixed delay period in milliseconds (ms). It takes values ranging from 0 to 6000000 ms. The default value is 500 ms.
• Random – random delay values within a chosen range (between start and stop), in ms. Both start and stop durations take values ranging from 0 to
Table 3-88. Variable Test GUI Operations
Operation Description
Opens a window that permits you to define an expression name (corresponding to an output) and an expression sequence to be evaluated at test execution time.
Edits an existing expression by opening the same window as that displayed by clicking the Add new
expression .
Performs common cut, copy paste operations.
Deletes an expression entry.
, Since the expression sequences order is relevant (higher ranked expressions are evaluated first), ordering the list is possible using these s.
Note: Evaluating an expression sequence is done by the Expression Evaluator, whose supported operations and operators are described in Appendix B, The Expression Evaluator Syntax.
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6000000 ms. The default values are random between start = 500 ms and stop = 5000 ms.
Procedure Description
Declares a procedure in the current test scenario. A procedure is used to encapsu-late several functions in a single script for further reuse.
Parameters
This function block has no parameters.
Exit Procedure Description
Determines the output of a procedure. A procedure may have one or more out-puts. Each Exit Procedure function represents a separate output for the Procedure function block.
Parameters
The optional output name. It is recommended to use this option and to give sug-gestive names to each output. Thus, you can make the difference between several outputs in the same procedure. The name entered here is the output name in the Procedure function block.
Counter Op Description
Inserts user-defined statistic counters, which can be incremented, decremented, or reset. The counter value can be subject to a post-execution analysis process.
Parameters
The user-defined statistic counters.
The list of user-defined statistic counters can be modified and different opera-tions can be applied to them:
• Increment
• Decrement
• Increment with expression
• Decrement with expression
• Set value (expression)
• Reset
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Test Time Description
Assesses the current time. It can be used to control the execution flow depending on time; the user can add more time intervals, each representing a new output for the function. The output is selected by comparing each time interval with the application’s local time. The Mismatch output is selected if no output includes the local time (and day, if used).
For a description of the Test Time function parameters, refer to Table 3-89.
Log Message Description
Enables you to add messages that are included in the execution log. These mes-sages may include variables that are substituted at runtime with the appropriate values.
Please note, however, that expressions including messages will not be fully eval-uated, only value substitution will be done. For example, a log message such as “This is $Iter+1” will not evaluate the $Iter+1 expression, but only substitute the value of the $Iter variable.
Parameters
The message to include in the log.
Dump Variables Description
Generates a log containing all variables from all engines and the user-defined variables. For engines not available for the channel containing this script func-tion, the variables have empty values.
Table 3-89. Test Time Parameters
Name Description
Time Interval (list)
The time interval to be compared with the local time. Its duration can span 1 second (s) to a whole week. There are two cases:
• Time interval in the same day, specified as two time limits, in hh:mm:ss format (for example, 15:23:32-23:34:55).
• Time interval in different days of the week, specified as two time limits in day_of_the_week hh:mm:ss format (for exam-ple, Sun 15:23:32-Mon 23:34:55).
Use Day Of The Week
If selected, the user can compare the time and days of the week. The default is Disabled.
Between … And …
The two-time stamps (and days of the week, if used) defining the time interval. You can specify the time in the hh:mm:ss AM/PM format.
NOTE: The time displays in the Time Interval list in 24h format.
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Parameters
N/A
Error Handler Description
This script function can be used to minimize the number of connectors in a script. If you include this function in a scenario and the other functions do not have the 'Error,' 'Timeout,' 'Disconnected,' or 'Invalid' outputs connected, the Error Han-dler function is automatically called.
Parameters
By double-clicking the function in the scenario editor, the Error Handler Proper-ties display, enabling you to access the parameters described in Table 3-90.
VoIP H323 RAS Library
The VoIP H.323 RAS Library includes the following script functions:
• H323 Register on page 3-116
• H323 UnRegister on page 3-117
H323 Register Registers an H323 endpoint with a gatekeeper.
Table 3-90. Error Handler - Parameters
Parameter Description Default Value
Properties Select the outputs to display (at least one) – choose from the following list of checkboxes:
• Error
• Timeout
• Disconnected
• Invalid
• Transport Failure
• Found
• Not Found
Enabled Error and Timeout checkboxes
Output Settings
Error – the function returns an internal error. Failed
Time-out – no function is running. Warning
Note: Make sure that there is a single Error Handler function per channel.
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Register Properties: Parameters
Table 3-91 describes the Register function parameters.
Register Properties: Output Settings
Table 3-92 describes the outputs available for the Register function.
H323 UnRegister Unregisters a currently registered H323 endpoint.
Table 3-91. H323 Register Properties - Parameters
Name Description
Delay Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Time to Live The time after which registration expiresd (milliseconds), specified as either of the following:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Delay Between Digits, in milliseconds (ms)
• PHONE_WAIT_TIME, in milliseconds (ms)
• MGCP Timeout, in milliseconds (ms)
Table 3-92. H.323 Register Properties - Output Settings
Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Rejected Rejects the endpoint request.
The default resolution for this output is SUCCESS.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
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H.323 Unregister: Parameters
Table 3-93 describes the Unregister function parameters.
H323 Unregister: Output Settings
Table 3-94 describes the outputs available for the Unregister function.
VoIP H323 Functions Library
The VoIP H.323 Test Library includes the following script functions:
• Make Call on page 3-118
• Receive Call on page 3-120
• End Call on page 3-122
Make Call Originates an H323 call to the specified endpoint alias or IP address.
Make Call Properties: Parameters
Table 3-95 describes the Make Call function parameters.
Table 3-93. H.323 Unregister Parameters - Title
Name Description
Delay The time to wait before the function starts. It can be used for synchronization reasons (that is, to synchronize the Make Call function with the Receive Call one). It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
The default is Static Expression, 0 ms.
Table 3-94. H.323 Unregister Parameters - Output Settings
Name Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
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Make Call Properties: Output Settings
Table 3-96 describes the Make Call Output Settings parameters.
Table 3-95. Make Call - Parameters
Name Description
Use Activity Settings If selected, the call destination alias and IP address are those specified at activity level.
Destination Alias When the Use Activity Settings option is not selected, this parameter specifies the destination phone alias.
This field supports the use of test scenario variables.
Destination IP Address
When the Use Activity Settings option is not selected, this parameter specifies the destination phone IP address (either IPv4 or IPv6).
This field supports the use of test scenario variables.
Delay before execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Delay Between Digits, in milliseconds (ms)
• PHONE_WAIT_TIME, in milliseconds (ms)
• MGCP Timeout, in milliseconds (ms)
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Receive Call Sets up the parameters for receiving an H323 call.
Receive Call Properties: Parameters
Table 3-97 describes the Receive Call function parameters.
Table 3-96. MakeCall - Output Settings
Properties Description
Output Settings
Ok A call was successfully established.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while trying to establish the call.
The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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Receive Call Properties: Output Settings
Table 3-98 describes the Receive Call Output Settings parameters.
Table 3-97. Receive Call - Parameters
Name Description
Delay before answer Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Timeout The time, in milliseconds (ms), that the script function waits for it to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Delay Between Digits, in milliseconds (ms)
• PHONE_WAIT_TIME, in milliseconds (ms)
• MGCP Timeout, in milliseconds (ms)
Table 3-98. Receive Call - Output Settings
Properties Description
Output Settings
Ok A call was successfully received.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while receiving the notification.
The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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End Call Ends an H323 call.
End Call Properties: Parameters
Table 3-99 describes the End Call function parameters.
End Call Properties: Output Settings
Table 3-100 describes the End Call Output Settings parameters.
Table 3-99. End Call - Parameters
Name Description
Delay before execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Wait other party to disconnect
If selected, waits for the configured amount of time (in millisecond) for the other end to disconnect the call.
This parameter can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Delay Between Digits, in milliseconds (ms)
• PHONE_WAIT_TIME, in milliseconds (ms)
• MGCP Timeout, in milliseconds (ms)
NOTE: If this option is not selected, the endpoint initiates the call disconnection.
Table 3-100. End Call - Output Settings
Properties Description
Output Settings
Ok A call was successfully ended.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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VoIP H248 Functions Library
The VoIP Megaco/H.248 Test Library includes the following script functions that emulate complete transactions, comprising both command requests and replies that manipulate contexts, terminations, events, and signals.
H.248 MGC Library • Add on page 3-123
• Modify on page 3-127
• Move on page 3-127
• Subtract on page 3-128
• AuditVal on page 3-128
• AuditCap on page 3-128
• SrvChange (MGC) on page 3-128
• Wait Notify on page 3-128
• Wait SrvChange (MGC) on page 3-133
• Wait Requests (MGC) on page 3-134
Add This function sends a transaction request with one or more Add commands and waits for the transaction reply. If the corresponding H248 activity is configured with the Use TransactionResponseAck option, the script function also sends a final acknowledgement.
Add: Tx Request
Table 3-101 describes the request parameters of the Add script function.
Table 3-101. Add Properties - Tx Request
Name Description
Context ID The request context ID, which can be any of the following:
• - : This denominates the NULL context.
• $ : This denominates an arbitrary context that is cho-sen by the device.
• * : This denominates all contexts.
• <AUTO> : This denominates the context associated with the current user.
Note: Numerical values and expressions (enclosed in [ ] ) are allowed.
Send ContextRequest
If enabled, a user-configurable context request is sent. To edit the request, select this option and click the Edit .
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Send ContextAtribAuditRequest
If enabled, a user-configurable context attributes audit request is sent. To edit the request, select this option and click the Edit .
For each command to be sent, a property page named Add Request#n is displayed and permits you to configure the following command parameters:
• The request termination ID, which can be any of the fol-lowing:
• ROOT : This denominates the ROOT termination.
• $ : This denominates an arbitrary termination that is chosen by the device.
• * : This denominates all terminations.
• <AUTO-PHYSICAL> : This denominates the physi-cal termination of the context associated with the current user.
• <AUTO-RTP1> : This denominates the first RTP ter-mination of the context associated with the current user.
• <AUTO-RTP2> : This denominates the second RTP termination of the context associated with the cur-rent user.
• The following options related to flags:
• Optional: If selected, the command’s optional flag is set.
• Wildcard Return: If selected, the command’s wildcard flag is set.
• A command-specific list of descriptors (only the descriptors that can be contained in the current com-mand) that are sent in the current command request.
When a descriptor is selected, the pane below displays an editable tree representation of the descriptor.
Refer to Appendix C, Using the H248 Descriptor Editor for more information on editing message descriptors.
Clicking the Add adds a new Add command and a corresponding property page as a separate tab (named Add Request#n).
Clicking the Delete deletes the current Add command property page.
You cannot delete if there is only one property page.
Clicking the Preview displays a window displaying the current transaction request code. The following options are available:
• Encode Type: Specifies the display mode as Normal, Compact, or Pretty.
• Version: Displays the selected protocol request ver-sion.
Table 3-101. Add Properties - Tx Request (Continued)
Name Description
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Add: Rx Reply
Table 3-102 describes the expected reply parameters of the Add script function.
Clicking the Templates opens a window that allows the import of a transaction from a list of existing templates.
Once a template is selected, you can view the contents of the template, the folder location, and a short description of the template.
Clicking the Import opens a window that allows the import of a H248 message in plain text format.
Table 3-102. Add Properties - Rx Reply
Name Description
Context ID The context ID of the expected reply, which can be any of the following:
• - : This denominates the NULL context.
• $ :This denominates an arbitrary context that is cho-sen by the device.
• * : This denominates all contexts.
• <AUTO> : This denominates the context associated with the current user.
• <ANY> : If selected, the context ID value is ignored.
Note: Numerical values and expressions (enclosed in [ ]) are allowed.
Expect message error
If selected, the expected reply message contains an error descriptor at message level. The expected error code may be specified in the adjacent field.
Expect transaction error
If selected, the expected reply message contains an error descriptor at transaction level. The expected error code may be specified in the adjacent field.
Expect action error If selected, the expected reply message contains an error descriptor at action level. The expected error code may be specified in the adjacent edit field.
Expect transaction/action reply
If selected, the expected reply message can be edited in the pane below.
Expect ContextReply
If enabled, context parameters following a context request are expected. The expected context parameters can be edited by clicking the Edit .
For each expected command reply, a property page (Add Reply#n) is displayed and permits you to edit the following reply parameters:
Table 3-101. Add Properties - Tx Request (Continued)
Name Description
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Add: Parameters
Table 3-9 describes the execution paramters of the Add script function .
• The termination ID of the reply, which can be any of the following:
• ROOT : This denominates the ROOT termination.
• $ : This denominates an arbitrary termination that is chosen by the device.
• * : This denominates all terminations.
• <AUTO-PHYSICAL> : This denominates the phys-ical termination of the context associated with the current user.
• <AUTO-RTP1> : This denominates the first RTP termination of the context associated with the cur-rent user.
• <AUTO-RTP2> : This denominates the second RTP termination of the context associated with the current user.
• <ANY> : If selected, the termination ID value is ignored.
• A command-specific list of descriptors (only the descriptors that can be contained in the current com-mand) that are expected in the current command reply.
When a descriptor is selected, the pane below displays an editable tree representation of the descriptor.
Refer to Appendix C, Using the H248 Descriptor Editor for more information on editing descriptors.
Table 3-102. Add Properties - Rx Reply (Continued)
Name Description
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Add: Output Settings
Table 3-104 describes the outputs available for the Add script function.
Modify This function sends a transaction request with one or more Modify commands and waits for the transaction reply for the specified timeout. If the corresponding H248 activity is configured with the Use TransactionResponseAck option, the script function also sends a final acknowledgement.
For a description on the configuration parameters, refer to Table 3-101 through Table 3-104.
Move Sends a transaction request with one or more Move commands and waits for the transaction reply for the specified timeout. If the corresponding H248 activity is
Table 3-103. Add Properties - Parameters
Name Description
Delay Before Execution
Delays the function execution by a duration that can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, Sleep 2000 in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
Timeout The time, in milliseconds (ms), that the script function waits for it to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Delay Between Digits, in milliseconds (ms)
• PHONE_WAIT_TIME, in milliseconds (ms)
• MGCP Timeout, in milliseconds (ms)
Table 3-104. Add Properties - Output Settings
Output Name
Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Timeout The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
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configured with the Use TransactionResponseAck option, the script function also sends a final acknowledgement.
For a description on the configuration parameters, refer to Table 3-101 through Table 3-104.
Subtract Sends a transaction request with one or more Subtract commands and waits for the transaction reply for the specified timeout. If the corresponding H248 activity is configured with the Use TransactionResponseAck option, the script function also sends a final acknowledgement.
For a description on the configuration parameters, refer to Table 3-101 through Table 3-104.
AuditVal Sends a transaction request with one or more AuditValue commands and waits for the transaction reply for the specified timeout. If the corresponding H248 activity is configured with the Use TransactionResponseAck option, the script function also sends a final acknowledgement.
For a description on the configuration parameters, refer to Table 3-101 through Table 3-104.
AuditCap Sends a transaction request with one or more AuditCapabilities commands and waits for the transaction reply for the specified timeout. If the corresponding H248 activity is configured with the Use TransactionResponseAck option, the script function also sends a final acknowledgement.
For a description on the configuration parameters, refer to Table 3-101 through Table 3-104.
SrvChange (MGC) Sends a transaction request with one or more ServiceChange commands and waits the transaction reply for the specified timeout. If the corresponding H248 activity is configured with the Use TransactionResponseAck option, the script function also sends a final acknowledgement.
For a description on the configuration parameters, refer Table 3-101 through Table 3-104.
Wait Notify Waits for a transaction request with one or more Notify commands for the speci-fied time (timeout value) and then sends a configured transaction reply. If the corresponding H248 activity is configured with the Use TransactionResponse-Ack option, the script function also waits for a final acknowledgement.
Wait Notify: Rx Request
Table 3-105 describes the request parameters of the Wait Notify script function.
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Table 3-105. Wait Notify Properties - Rx Request
Name Description
Context ID The request context ID, which can be any of the following:
• - : This denominates the NULL context.
• $ : This denominates an arbitrary context that is cho-sen by the device.
• * : This denominates all contexts.
• <AUTO> : This denominates the context associated with the current user.
• <ANY> : When selected, the context ID value is ignored.
Note: Numerical values and expressions (enclosed in [ ] ) are allowed.
Expect ContextRequest
If enabled, a user-configurable context request is expected. To edit the expected request, select this option and click the Edit .
Expect ContextAtribAuditRequest
If enabled, a user-configurable context attributes audit request is expected. To edit the expected request, select this option and click the Edit .
For each expected command request, a property page named Notify Request#n is displayed and permits you to configure the following command parameters:
• The command termination ID, which can be any of the following:
• ROOT : This denominates the ROOT termination.
• $ : This denominates a termination that is chosen by the device.
• * : This denominates all terminations.
• <AUTO-PHYSICAL> : This denominates the current physical termination of the context associated with the current user.
• <AUTO-RTP1> : This denominates the first RTP ter-mination of the context associated with the current user.
• <AUTO-RTP2> : This denominates the second RTP termination of the context associated with the cur-rent user.
• <ANY> : When selected, the termination ID value is ignored.
• The following command options:
• Optional: If selected, the command’s optional flag is set.
• Wildcard Return: If selected, the command’s wildcard flag is set.
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Wait Notify: Tx Reply
Table 3-106 describes the reply parameters of the Wait Notify script function.
• A command-specific list of descriptors (only the descriptors that can be contained in the current com-mand) that are expected in the current command request.
When a descriptor is selected, the pane below displays an editable tree representation of the descriptor.
Refer to Appendix C, Using the H248 Descriptor Editor for more information on editing message descriptors.
Clicking the Add adds a new expected command request and a corresponding property page as a separate tab (named Notify Request#n).
Clicking the Delete deletes the current command property page.
You cannot delete if there is only one property page.
Clicking the Preview displays a window displaying the current transaction code. The following options are available:
• Encode Type: Specifies the display mode as Normal, Compact, or Pretty.
• Version: Displays the selected protocol request ver-sion.
Clicking the Templates opens a window that allows the import of a transaction from a list of existing templates.
Once a template is selected, you can view the contents of the template, the folder location, and a short description of the template.
Clicking the Import opens a window that allows the import of a H248 message in plain text format.
Table 3-105. Wait Notify Properties - Rx Request (Continued)
Name Description
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Table 3-106. Wait Notify Properties - Tx Reply
Name Description
Context ID The reply context ID, which can be any of the following:
• - : This denominates the NULL context.
• $ : This denominates an arbitrary context that is cho-sen by the device.
• * : This denominates all contexts.
• <AUTO> : This denominates the context associated with the current user.
Note: Numerical values and expressions (enclosed in [ ]) are allowed.
Send message error If selected, the sent reply message contains an error descriptor at message level. The error code may be specified in the adjacent field.
Send transaction error
If selected, the sent reply message contains an error descriptor at transaction level. The sent error code may be specified in the adjacent field.
Send action error If selected, the sent reply message contains an error descriptor at action level. The sent error code may be specified in the adjacent edit field.
Send transaction/action reply
If selected, the reply message can be edited in the pane below.
Send ContextReply If enabled, the reply contains the requested context parameters. To edit the sent context parameters, select this option and click the Edit .
When the Send transaction/action reply option is selected, the reply commands can be configured individually using the GUI. For each command reply sent, a property page (Notify Reply#n) is displayed and permits you to edit the following parameters:
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Wait Notify: Parameters
Table 3-107 describes the execution parameters of the Wait Notify script func-tion.
• The reply terminationID, which can be any of the fol-lowing:
• ROOT : This denominates the ROOT termination.
• $ : This denominates an arbitrary termination that is chosen by the device.
• * : This denominates all terminations.
• <AUTO-PHYSICAL> : This denominates the phys-ical termination of the context associated with the current user.
• <AUTO-RTP1> : This denominates the first RTP termination of the context associated with the cur-rent user.
• <AUTO-RTP2> : This denominates the second RTP termination of the context associated with the current user.
• A command-specific list of descriptors (the descrip-tors that can be contained in the current command) that are to be included in the command reply.
When a descriptor is selected, the pane below displays an editable tree representation of the current descriptor.
Refer to Appendix C, Using the H248 Descriptor Editor for more information on editing descriptors.
Table 3-106. Wait Notify Properties - Tx Reply (Continued)
Name Description
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Wait Notify: Output Settings
Table 3-108 describes the outputs parameters of the Wait Notify script function.
Wait SrvChange (MGC)
Waits for a transaction request with one or more ServiceChange commands for the specified timeout and then sends a transaction reply. If the corresponding H248 activity is configured with the Use TransactionResponseAck option, the script function also waits for a final acknowledgement.
For a description on the configuration parameters, refer Table 3-105 through Table 3-108.
Table 3-107. Wait Notify Properties - Parameters
Name Description
Delay Before Execution
Delays the function execution by a duration that can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, Sleep 2000 in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
Timeout The time, in milliseconds (ms), that the script function waits for it to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled. It can be specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Delay Between Digits, in milliseconds (ms)
• PHONE_WAIT_TIME, in milliseconds (ms)
• MGCP Timeout, in milliseconds (ms)
Table 3-108. Wait Notify Properties - Output Settings
Output Name
Description
OK The function completed successfully.
The default resolution for this output is SUCCESS.
Timeout The default resolution for this output is WARNING.
Error The function has returned an internal error.
The default resolution for this output is FAILED.
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Wait Requests (MGC)
Waits for a transaction request with one or more specified H.248 commands for the specified timeout. If the transaction matches one of the expected commands, it sends the configured transaction reply and exits on the corresponding output. If a transaction request is received, but it doesn't match any of the templates, the 'Mismatched' output is used.
If the corresponding H248 activity is configured with the Use TransactionRe-sponseAck option, the script function also waits for a final acknowledgement.
For a description of the expected command configuration, refer to Table 3-105 and Table 3-106.
Note: Wait-type script functions accept messages with H.248 protocol combinations that are not allowed in practice (such as, for example, Subtract from a NULL context, Add with a NULL context) or combinations that are not valid in the current state of the gateway (such as, for example, mixing unrelated terminations and contexts).
Note: Wait-type script functions accept messages with H.248 protocol combinations that are not allowed in practice (such as, for example, Subtract from a NULL context, Add with a NULL context) or combinations that are not valid in the current state of the gateway (such as, for example, mixing unrelated terminations and contexts).
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H.248 MGW Library • Notify on page 3-135
• SrvChange (MGW) on page 3-135
• Wait Add on page 3-135
• Wait Modify on page 3-136
• Wait Move on page 3-136
• Wait Subtract on page 3-136
• Wait AuditVal on page 3-136
• Wait AuditCap on page 3-137
• Wait SrvChange (MGW) on page 3-137
• Wait Requests (MGW) on page 3-137
Notify Sends a transaction request with one or more Notify commands and waits for the transaction reply (replies) for the specified timeout. A Notify command allows the Media Gateway to inform the Media Gateway Controller of the occurrence of events in the Media Gateway.
For description on the configuration parameters, refer Table 3-101 through Table 3-104.
SrvChange (MGW) Sends a transaction request with one or more ServiceChange commands and waits the transaction reply for the specified timeout. A ServiceChange command allows the MG to notify the MGC that a termination or group of terminations is about to be taken out of service or has just been returned to service. Servi-ceChange commands are also used by the MG to announce its availability to a MGC (registration), and to notify the MGC of impending or completed restart of the MG.
For description on the configuration parameters, refer Table 3-101 through Table 3-104.
Wait Add Waits for a transaction request with one or more Add commands for the specified timeout and then sends a transaction reply. An Add command adds a termination to a context. An Add command on the first termination in a context is used to cre-ate a context.
For description on the configuration parameters, refer Table 3-105 through Table 3-108.
Note: Wait-type script functions accept messages with H.248 protocol combinations that are not allowed in practice (such as, for example, Subtract from a NULL context, Add with a NULL context) or combinations that are not valid in the current state of the gateway (such as, for example, mixing unrelated terminations and contexts).
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Wait Modify Waits for a transaction request with one or more Modify commands for the spec-ified timeout and then sends a transaction reply. A Modify command modifies the properties, events and signals of a termination.
For description on the configuration parameters, refer Table 3-105 through Table 3-108.
Wait Move Waits for a transaction request with one or more Move commands for the speci-fied timeout and then sends a transaction reply. A Move command atomically moves a termination from one context to another.
For description on the configuration parameters, refer Table 3-105 through Table 3-108.
Wait Subtract Waits for a transaction request with one or more Subtract commands for the specified timeout and then sends a transaction reply. A Subtract command dis-connects a termination from its context and returns statistics on the termination's participation in the context. The Subtract command on the last termination in a context deletes the context.
For description on the configuration parameters, refer Table 3-105 through Table 3-108.
Wait AuditVal Waits for a transaction request with one or more AuditValue commands for the specified timeout and then sends a transaction reply. An AuditValue command returns the current state of properties, events, signals and statistics of termina-tions.
For description on the configuration parameters, refer Table 3-105 through Table 3-108.
Note: Wait-type script functions accept messages with H.248 protocol combinations that are not allowed in practice (such as, for example, Subtract from a NULL context, Add with a NULL context) or combinations that are not valid in the current state of the gateway (such as, for example, mixing unrelated terminations and contexts).
Note: Wait-type script functions accept messages with H.248 protocol combinations that are not allowed in practice (such as, for example, Subtract from a NULL context, Add with a NULL context) or combinations that are not valid in the current state of the gateway (such as, for example, mixing unrelated terminations and contexts).
Note: Wait-type script functions accept messages with H.248 protocol combinations that are not allowed in practice (such as, for example, Subtract from a NULL context, Add with a NULL context) or combinations that are not valid in the current state of the gateway (such as, for example, mixing unrelated terminations and contexts).
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Wait AuditCap Waits for a transaction request with one or more AuditCapabilities commands for the specified timeout and then sends a transaction reply. An AuditCapabilities command returns all the possible values for the termination properties, events and signals allowed by the Media Gateway.
For description on the configuration parameters, refer Table 3-105 through Table 3-108.
Wait SrvChange (MGW)
Waits for a transaction request with one or more ServiceChange commands for the specified timeout and then sends a transaction reply. A MGC may announce a handover to the MG by sending it a ServiceChange Command. The MGC may also use ServiceChange to instruct the MG to take a termination or group of ter-minations in or out of service.
For description on the configuration parameters, refer Table 3-105 through Table 3-108.
Wait Requests (MGW)
Waits for a transaction request with one or more H.248 commands for the speci-fied timeout. If the transaction matches one of the expected requests, then it sends the configured transaction reply and exits on the corresponding output. If a transaction request is received but it doesn't match any of the templates, the 'Mis-matched' output is used.
For a description of the expected command configuration, refer to Wait Requests (MGC) on page 3-134.
Note: Wait-type script functions accept messages with H.248 protocol combinations that are not allowed in practice (such as, for example, Subtract from a NULL context, Add with a NULL context) or combinations that are not valid in the current state of the gateway (such as, for example, mixing unrelated terminations and contexts).
Note: Wait-type script functions accept messages with H.248 protocol combinations that are not allowed in practice (such as, for example, Subtract from a NULL context, Add with a NULL context) or combinations that are not valid in the current state of the gateway (such as, for example, mixing unrelated terminations and contexts).
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VoIP MGCP Functions Library
The MGCP Test Library includes the following script functions:
• Send NTFY on page 3-139
• Send DLCX (GW) on page 3-140
• Send RSIP on page 3-141
• Wait CRCX on page 3-141
• Wait DLCX (GW) on page 3-142
• Wait MDCX on page 3-142
• Wait RQNT on page 3-143
• Wait AUEP on page 3-143
• Wait AUCX on page 3-144
• Wait EPCF on page 3-144
• Wait Any Command (GW) on page 3-145
• Wait Command (GW) on page 3-146
• Send RQNT on page 3-148
• Send CRCX on page 3-149
• Send DLCX (CA) on page 3-149
• Send MDCX on page 3-150
• Send AUCX on page 3-150
• Send AUEP on page 3-151
• Send EPCF on page 3-151
• Wait NTFY on page 3-152
• Wait DLCX (CA) on page 3-152
• Wait Command (CA) on page 3-153
• Wait Any Command (CA) on page 3-153
• Wait RSIP on page 3-153
Overview Each script function implements an MGCP transaction, i.e. an MGCP command followed by an expected response.
A script function typically represents a sent MGCP command and the expected response message (for Send type functions), the corresponding transaction being closed when a matching response message is received.
MGCP MGW Functions
The following script functions are available:
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Send NTFY
The MGCP Notify command is used by a MGW to send notifications of the observed events when a triggering event occurs.
The Send Notify script function implements the transaction initiated by a Notify command.
Send Notify Properties:Tx Command
This page specifies the Notify command parameters. See Tx Command Page on page D-2.
Send Notify Properties:Rx Response
This page specifies the awaited response message following the sending of a Notify command. See Rx Response Page on page D-3.
Send Notify Properties:Parameters
Table 3-109 describes the Send Notify function parameters.
Send Notify Properties:Output Settings
Table 3-110 describes the Send Notify output parameters.
Table 3-109. Send Notify - Parameters
Name Description
Delay before execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
Transaction timeout
The time, in milliseconds (ms), that the script function waits for it to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
It can be configured as either of the following:
• A static user-defined value that is entered into the field.
• A random value between two user-specified values
• The timeout value configured at the activity level (Use activity settings option).
Table 3-110. Send Notify - Output Settings
Properties Description
200 OK The Send Notify function was executed successfully. The default resolution for this output is SUCCESS.
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Send DLCX (GW)
In some rare circumstances, a MGW may have to clear a connection, for example because it has lost the resource associated with the connection, or because it has detected that the endpoint no longer is capable or willing to send or receive media. In such situations, the MGW may then terminate the connection by using an MGCP DeleteConnection command.
The Send DLCX script function implements the transaction initiated by a sent DeleteConnection command.
Send DLCX Properties:Tx Command
This page specifies the DLCX command parameters. See Tx Command Page on page D-2.
Send Notify Properties:Rx Response
This page specifies the awaited response message following the sending of a DLCX command. See Rx Response Page on page D-3.
Send DLCX Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send DLCX Properties:Output Settings
Table 3-110 describes the Send DLCX Output parameters.
Timeout A timeout occurred while executing the Send Notify function. The default resolution for this output is WARNING.
Error The Send Notify function execution has returned an error. The default resolution for this output is FAILED.
Table 3-111. Send DLCX - Output Settings
Properties Description
250 Connection Deleted
The Send DLCX function was executed successfully. The default resolution for this output is SUCCESS.
Timeout A timeout occurred while executing the Send DLCX function. The default resolution for this output is WARNING.
Error The Send DLCX function execution has returned an error. The default resolution for this output is FAILED.
Table 3-110. Send Notify - Output Settings
Properties Description
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Send RSIP
The Send RSIP script function implements the transaction initiated by a RestartInProgress (RSIP) command sent from the MGW to the CA.
Send RSIP Properties:Tx Command
This page specifies the RSIP command parameters. See Tx Command Page on page D-2.
Send RSIP Properties:Rx Response
This page specifies the awaited response message following the sending of a RSIP command. See Rx Response Page on page D-3.
Send RSIP Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send RSIP Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait CRCX
The Wait CRCX script function implements the transaction initiated by the receiving of a CreateConnection (CRCX) command, followed by the sending of a response to this command.
Wait CRCX Properties:Rx Command
This page specifies the parameters of a received CRCX command. See Rx Command Page on page D-4.
Wait CRCX Properties:Tx Response
This page specifies the generated response message following the receiving of a CRCX command. See Tx Response Page on page D-4.
Wait CRCX Properties:Parameters
This page specifies the function execution parameters described in Table 3-115.
Wait CRCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
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Wait DLCX (GW)
The Wait DLCX script function implements the transaction initiated by the receiving of a DeleteConnection (CRCX) command, followed by the sending of a response to this command.
Wait DLCX Properties:Rx Command
This page specifies the parameters of a received DLCX command. See Rx Command Page on page D-4.
Wait DLCX Properties:Tx Response
This page specifies the generated response message following the receiving of a DLCX command. See Tx Response Page on page D-4.
Wait DLCX Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
Wait DLCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait MDCX
The Wait MDCX script function implements the transaction initiated by a the receiving of a ModifyConnection (CRCX) command, followed by the sending of a response to this command.
Wait MDCX Properties:Rx Command
This page specifies the parameters of a received MDCX command. See Rx Command Page on page D-4.
Wait MDCX Properties:Tx Response
This page specifies the generated response message following the receiving of a MDCX command. See Tx Response Page on page D-4.
Wait MDCX Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
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Wait MDCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait RQNT
The Wait RQNT script function implements the transaction initiated by the receiving of a RequestNotify (RQNT) command, followed by the sending of a response to this command.
Wait RQNT Properties:Rx Command
This page specifies the parameters of a received RQNT command. See Rx Command Page on page D-4.
Wait RQNT Properties:Tx Response
This page specifies the generated response message following the receiving of a RQNT command. See Tx Response Page on page D-4.
Wait RQNT Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
Wait RQNT Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait AUEP
The Wait AUEP script function implements the transaction initiated by the receiving of an AuditEndpoint (AUEP) command, followed by the sending of a response to this command.
Wait AUEP Properties:Rx Command
This page specifies the parameters of a received AUEP command. See Rx Command Page on page D-4.
Wait AUEP Properties:Tx Response
This page specifies the generated response message following the receiving of an AUEP command. See Tx Response Page on page D-4.
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Wait AUEP Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
Wait AUEP Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait AUCX
The Wait AUCX script function implements the transaction initiated by the receiving of an AuditConnection (AUCX) command, followed by the sending of a response to this command.
Wait AUCX Properties:Rx Command
This page specifies the parameters of a received AUCX command. See Rx Command Page on page D-4.
Wait AUCX Properties:Tx Response
This page specifies the generated response message following the receiving of an AUCX command. See Tx Response Page on page D-4.
Wait AUCX Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
Wait AUCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait EPCF
The Wait EPCF script function implements the transaction initiated by the receiving of an EndpointConfiguration (EPCF) command, followed by the send-ing of a response to this command.
Wait EPCF Properties:Rx Command
This page specifies the parameters of a received EPCF command. See Rx Command Page on page D-4.
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Wait EPCF Properties:Tx Response
This page specifies the generated response message following the receiving of an EPCF command. See Tx Response Page on page D-4.
Wait EPCF Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
Wait EPCF Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait Any Command (GW)
The Wait Any Command (GW) command waits for any MGCP command with a user-specified set of parameters.
Wait Any Command (GW) Properties:Rx Command
This page specifies the parameters of a received MGCP command. See Rx Command Page on page D-4.
Wait Any Command (GW) Properties:Tx Response
This page specifies the generated response message following the receiving of a MGCP command. See Tx Response Page on page D-4.
Wait Any Command (GW) Properties: Parameters
Table 3-112 describes the Wait Any function parameters.
Table 3-112. Wait Any Command - Parameters
Name Description
Delay before execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
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Wait Any Command (GW) Properties: Output Settings
Table 3-113 describes the Wait Any Output parameters.
Wait Command (GW)
The Wait Command (GW) function specifies one or more MGCP commands awaited by the MGW. If any of these commands is received, the user-configured response to the command is sent and, if executed successfully, the function exits on the output identifying the matched command.
Wait Command Properties: Templates
Table 3-114 describes the request parameters of the Wait Command (GW) script function.
Timeout The time, in milliseconds (ms), that the script function waits for it to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
It can be configured as either of the following:
• A static user-defined value that is entered into the field.
• A random value between two user-specified values
• The timeout value configured at the activity level (Use activ-ity settings option).
Transaction delay
The time, in milliseconds (ms), that the script function waits before executing.
It can be configured as either of the following:
• A static user-defined value that is entered into the field.
• A random value between two user-specified values.
Table 3-113. Wait Any- Output Settings
Properties Description
OK The Wait Any function was executed successfully. The default resolution for this output is SUCCESS.
Timeout A timeout occurred while executing the Wait Any function. The default resolution for this output is WARNING.
Error The Wait Any function execution has returned an error. The default resolution for this output is FAILED.
Table 3-112. Wait Any Command - Parameters (Continued)
Name Description
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Wait Command Properties: Parameters
Table 3-115 describes the Wait Command function parameters.
Table 3-114. Wait Command (GW) Properties - Templates
Name Description
A list of commands to be matched by the MGW. If any of the received commands is matched, the function exits on the corresponding output.
For example, considering we have a configured list such as the following, if an MDCX-type command with matching parameters is received, then the Wait Command function sends the 200 Ok response code and exits on the MDCX1 output that was automatically created when adding the command to the list.
Clicking the Add adds a new command to be matched by the MGW. For each added command, the awaited MGCP command and parameters, as well as the response can be configured as described in Appendix D, Using the MGCP Parameter Editor.
Clicking the Delete deletes the currently selected command from the list of awaited commands. You cannot delete the last remaining command.
Clicking the Edit displays a window that permits you to edit the command parameters and the command response.
Table 3-115. Wait Command - Parameters
Name Description
Delay before execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
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Wait Command Properties: Output Settings
In addition to the common Timeout and Error outputs available for all script functions, the Wait Command function provides additional outputs for all awaited messages configured in the function. These additional outputs are auto-matically created when adding a command to the list of commands to be matched.
For example, assuming we had configured the list of commands shown in Table 3-1, the additional MDCX1 and AUCX1 outputs would become available in the script function.
Figure 3-1. Wait Command Function - Configured Command List
MGCP CA Functions
The following functions are available:
Send RQNT
The Send RQNT script function implements the transaction initiated by a RequestNotify (RQNT) command sent by the CA.
Send RQNT Properties:Tx Command
This page specifies the RQNT command parameters. See Tx Command Page on page D-2.
Timeout The time, in milliseconds (ms), that the script function waits for it to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
It can be configured as either of the following:
• A static user-defined value that is entered into the field.
• A random value between two user-specified values
• The timeout value configured at the activity level (Use activ-ity settings option).
Transaction delay
The time, in milliseconds (ms), that the script function waits before executing.
It can be configured as either of the following:
• A static user-defined value that is entered into the field.
• A random value between two user-specified values.
Table 3-115. Wait Command - Parameters (Continued)
Name Description
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Send RQNT Properties:Rx Response
This page specifies the awaited response message following the sending of an RQNT command. See Rx Response Page on page D-3.
Send RQNT Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send RQNT Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Send CRCX
The Send CRCX script function implements the transaction initiated by a Cre-ateConnection (CRCX) command sent by the CA.
Send CRCX Properties:Tx Command
This page specifies the CRCX command parameters. See Tx Command Page on page D-2.
Send CRCX Properties:Rx Response
This page specifies the awaited response message following the sending of an CRCX command. See Rx Response Page on page D-3.
Send CRCX Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send CRCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Send DLCX (CA)
The Send DLCX script function implements the transaction initiated by a DeleteConnection (DLCX) command sent by the CA.
Send DLCX Properties:Tx Command
This page specifies the DLCX command parameters. See Tx Command Page on page D-2.
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Send DLCX Properties:Rx Response
This page specifies the awaited response message following the sending of an DLCX command. See Rx Response Page on page D-3.
Send DLCX Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send DLCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Send MDCX
The Send MDCX script function implements the transaction initiated by a Mo-difyConnection (MDCX) command sent by the CA.
Send MDCX Properties:Tx Command
This page specifies the MDCX command parameters. See Tx Command Page on page D-2.
Send MDCX Properties:Rx Response
This page specifies the awaited response message following the sending of an MDCX command. See Rx Response Page on page D-3.
Send MDCX Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send MDCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Send AUCX
The Send AUCX script function implements the transaction initiated by an AuditConnection (AUCX) command sent by the CA.
Send AUCX Properties:Tx Command
This page specifies the AUCX command parameters. See Tx Command Page on page D-2.
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Send AUCX Properties:Rx Response
This page specifies the awaited response message following the sending of an AUCX command. See Rx Response Page on page D-3.
Send AUCX Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send AUCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Send AUEP
The Send AUEP script function implements the transaction initiated by a Audit Connection (AUEP) command sent by the CA.
Send AUEP Properties:Tx Command
This page specifies the AUEP command parameters. See Tx Command Page on page D-2.
Send AUEP Properties:Rx Response
This page specifies the awaited response message following the sending of an AUEP command. See Rx Response Page on page D-3.
Send AUEP Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send AUEP Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Send EPCF
The Send EPCF script function implements the transaction initiated by a End-point Configuration (EPCF) command sent by the CA.
Send EPCF Properties:Tx Command
This page specifies the EPCF command parameters. See Tx Command Page on page D-2.
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Send EPCF Properties:Rx Response
This page specifies the awaited response message following the sending of an EPCF command. See Rx Response Page on page D-3.
Send EPCF Properties:Parameters
This page specifies the function execution parameters described in Table 3-109.
Send EPCF Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait NTFY
The Wait NTFY script function implements the transaction initiated by the receiving of an MGCP Notify command sent by an MGW.
Wait NTFY Properties:Rx Command
This page specifies the parameters of a received NTFY command. See Rx Command Page on page D-4.
Wait NTFY Properties:Tx Response
This page specifies the generated response message following the receiving of an NTFY command. See Tx Response Page on page D-4.
Wait NTFY Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
Wait NTFY Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait DLCX (CA)
The Wait DLCX script function implements the transaction initiated by the receiving at the CA of an MGCP DeleteConnection (DCLX) command sent by an MGW.
Wait DLCX Properties:Rx Command
This page specifies the parameters of a received DLCX command. See Rx Command Page on page D-4.
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Wait DLCX Properties:Tx Response
This page specifies the generated response message following the receiving of an DLCX command. See Tx Response Page on page D-4.
Wait DLCX Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
Wait DLCX Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
Wait Command (CA)
The Wait Command (CA) script function specifies one or more MGCP com-mands awaited by the CA. If any of these commands is received, the user-config-ured response to the command is sent and, if executed successfully, the function exits on the output identifying the matched command.
See Wait Any Command (GW) on page 3-145.
Wait Any Command (CA)
The Wait Any Command (CA) command waits for any MGCP command with a user-specified set of parameters.
See Wait Command (GW) on page 3-146.
Wait RSIP
The Wait RSIP script function implements the transaction initiated by the receiving at the CA of an MGCP RestartInProgress (RSIP) command sent by an MGW.
Wait RSIP Properties:Rx Command
This page specifies the parameters of a received RSIP command. See Rx Command Page on page D-4.
Wait RSIP Properties:Tx Response
This page specifies the generated response message following the receiving of an RSIP command. See Tx Response Page on page D-4.
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Wait RSIP Properties:Parameters
This page specifies the function execution parameters, the same as those described in Table 3-115.
Wait RSIP Properties:Output Settings
This page specifies the function output parameters, the same as those described in Table 3-110.
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Digital T1/E1 Functions Library
The Digital T1/E1 Test Library includes the following script functions:
• Make Call on page 3-155
• Receive Call on page 3-157
• End Call on page 3-158
• Path Confirmation on page 3-159
• Talk on page 3-163
• Listen on page 3-165
• Voice Session on page 3-166
• Generate DTMF on page 3-167
• Detect DTMF on page 3-169
• Generate MF on page 3-171
• Detect MF on page 3-171
• Generate Tone on page 3-171
• Wait for Tone on page 3-173
Make Call This script function initiates a call to the specified destination.
Make Call Properties: Parameters
Table 3-116 describes the Make Call function parameters.
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Make Call Properties: Output Settings
Table 3-117 describes the Make Call output parameters.
Table 3-116. Make Call - Parameters
Name Description
Delay Before Execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
No Answer Timeout The time, in milliseconds (ms), that the script function waits for the function to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
This parameter can be specified as either of the following:
• A user-defined value that is entered into the field.
• As the Global value, in which case the value is used configured in the IxLoad Global Settings window.
Destination Number The destination number for the call originated by the PSTNDigital activity, which can be specified as either of the following:
• Use Dest. Phone Value: The destination number is taken from the activity’s dial plan configuration
• Custom: A user-configured value.
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Receive Call This script function answers an incoming call.
Receive Call Properties: Parameters
Table 3-118 describes the Receive Call function parameters.
Table 3-117. MakeCall - Output Settings
Properties Description
Output Settings
Connected A call was successfully established.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while trying to establish the call. The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication. The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-118. Receive Call - Parameters
Name Description
Delay before answer Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
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Receive Call Properties: Output Settings
Table 3-119 describes the Receive Call output parameters.
End Call This script function terminates an established call.
End Call Properties: Parameters
Table 3-120 describes the End Call function parameters.
Wait for call The time, in milliseconds (ms), that the script function waits for it to execute. If this time interval terminates without the function being executed, the Timeout function output is enabled.
This parameter can be specified as either of the following:
• A user-defined value that is entered into the field.
• As the Global value, in which case the value is used configured in the IxLoad Global Settings window.
Reject call When a call is rejected by the Receive Call function, this parameter can be used for providing a call reject reason.
This parameter can be configured to one of the predefined reasons, or to a Global value, in which case the used value is that configured in the IxLoad Global Settings window.
Table 3-119. Receive Call - Output Settings
Properties Description
Output Settings
Connected A call was successfully received.
The default resolution for this output is SUCCESS.
Timeout A timeout occurred while receiving the call.
The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication.
The default resolution for this output is WARNING.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
Table 3-118. Receive Call - Parameters (Continued)
Name Description
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End Call Properties: Output Settings
Table 3-100 describes the End Call output parameters.
Path Confirmation This script function executes a Path Confirmation sequence, wherein the path confirmation initiator sends a specific digit (DTMF/MF/tone) sequence and then waits to receive another digit (DTMF/MF/Tone) sequence from the remote party.
Path Confirmation Properties: Parameters
Table 3-122 describes the Path Confirmation function parameters.
Table 3-120. End Call - Parameters
Name Description
Delay before execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Wait other party to disconnect
If selected, waits for the configured amount of time (in milliseconds) for the other party to disconnect the call. When this option is not selected, the End Call script function initiates the call disconnection itself.
When this option is selected, the Wait duration parameter can be specified as either of the following:
• A user-defined value that is entered into the filed.
• As the Global value, in which case the value is used configured in the Global Settings window.
Table 3-121. End Call - Output Settings
Properties Description
Output Settings
Ok A call was successfully ended.
The default resolution for this output is SUCCESS.
Error The function has returned an error following an inexistent call handle, inexistent line, or erroneous parameters evaluation condition.
The default resolution for this output is FAILED.
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Table 3-122. Path Confirmation - Parameters
Name Description
Delay Before Execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Path Confirmation Method
Selects the operating mode as either of the following:
• Simplex (default): Transmission and reception of the path confirmation sequence is done in an alternating manner.
• Duplex: Transmission and reception of the path con-firmation sequence is done in an alternating manner.
• Synchronized Duplex (VoIP Style): Transmission and reception of the path confirmation sequence is done in both directions simultaneously.
Specified Digit Sequence of...
Selects the mode – DTMFs, MFs, or tones – in which the path confirmation operation is performed:
• Specified Digit Sequence of DTMFs - The path confir-mation initiator sends the specified DTMF digits and expects to receive the same DMTF digits. The path confirmation receiver expects the specified DTMF digits and then sends them back.
• Specified Digit Sequence of MFs - The path confirma-tion initiator sends the specified MF digits and expects to receive the same MF digits. The path confirmation receiver expects the specified MF digits and then sends them back.
• Specified Sequence of Custom Tones - The path con-firmation initiator sends the specified custom tone sequence and expects to receive the same sequence. The path confirmation receiver expects the specified custom tone sequence and then sends it back.
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Path Confirmation Properties: Tone Detection /Generation
Table 3-123 describes the Path Confirmation Tone Detection / Generation parameters.
Time Guard Delay This parameter specifies the time used as guard for path confirmation that supports the separation of Tx/Rx paths in simplex mode, and separation of phases of sequences for duplex modes. It can be a value in ms with valid values between 0 (default) and 60000 ms).
Start Mode Specifies how the channels running this script function start the path confirmation sequence:
• Initiate - the channel(s) executing this script function first send the path confirmation digits and then wait, and so on (alternating).
• Wait for - the channel(s) executing this script function first wait for the path confirmation digits and then send the digits, and so on (alternating).
• Auto - the channel(s) executing this script function rely on signaling to decide if they wait for or send the digit sequence. The path confirmation initiator is the party that receives the call. This is the default option.
Note: When a DTMF/MF/Tone sequence is specified using expressions, you must ensure that every two pair channels evaluate the expressions to the same value, otherwise the path confirmation sequence fails; the path confirmation initiator/receiver sends and expects the evaluated expression (interpreted as a digit sequence).
Table 3-122. Path Confirmation - Parameters (Continued)
Name Description
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Path Confirmation Properties: Advanced Settings
Table 3-124 describes the Path Confirmation Advanced Settings parameters.
Table 3-123. Path Confirmation - Tone Detection / Generation
Name Description
DTMF/MF Generation
Specifies the following DTMF/MF generation parameters using custom values, or by applying the values specified in the Global Settings | T1/E1 | Voice window when the Use values from Global Settings option is selected:
• Tone Duration: The time (in ms) required for a single tone (DTMF/MF/Custom Tone) to be generated. The range of values is 50 through 10000 ms. The default value is 200 ms.
• Inter Tone Interval: The maximum amount of time (in ms) between two consecutive generated DTMFs/MFs/Custom Tones. The range of values is 50 to 10000 ms. The default value is 200 ms.
• Signal Power: The attenuation (in dBm) of the DTMF/MF. The minimum attenuation is -54 dBm and the maximum is -3 dBm. The default value is -10 dBm.
DTMF/MF/Tone Detection
Specifies the following DTMF/MF/Tone detection parameters using custom values, or by applying the values specified in the Global Settings | T1/E1 | Voice window when the Use values from Global Settings option is selected:
• Maximum delay between tones: The maximum amount of time (in ms) allowed between consecutive digits (DTMFs/MFs/custom tones) for a proper detec-tion. After this period elapses, the function exits on the Timeout output. The range of values is 0 to 65520 ms. The default value is 500 ms.
• First Tone Timeout: The time (in ms) allowed for detecting the first digit (DTMF/MF/custom tone). After this period elapses, the function exits on Timeout out-put. The range of values is 0 to 65520 ms. The default value is 500 ms.
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Path Confirmation Properties: Output Settings
Table 3-125 describes the Path Confirmation Output Settings parameters.
Talk This script function plays back a wave file or a QoV clip from the IxLoad Resource Pool.
Talk Properties: Parameters
Table 3-126 describes the Talk function parameters.
Table 3-124. Path Confirmation - Advanced Settings
Name Description
Playback Specifies the DTMF/MF/tone playback mode using either of the options below, or by applying the values specified in the Global Settings | T1/E1 Voice window (for the Use Global Setting option enabled):
• Play: Specifies the number of times to play.
• Repeat Continuously for: Specify the period of time to play. The default value is 1000 seconds.
• Use Talk Time (for all objectives except Channels): Plays the DTMF/MF/Tone sequence for the duration of the Talk Time call parameter.
Table 3-125. Path Confirmation - Output Settings
Properties Description
Output Settings
Connected The path confirmation sequence was correctly executed for the specified number of times or duration.
The default resolution for this output is SUCCESS.
Timeout The path confirmation sequence failed due to a timeout (e.g. DTMF/MF/Tone detection timeout).
The default resolution for this output is WARNING.
Disconnected On the established connection, the remote party dropped the call by sending a disconnect indication. The default resolution for this output is WARNING.
Error The function has returned an error.
The default resolution for this output is FAILED.
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Talk Properties: Advanced Settings
Table 3-127 describes the Talk function Advanced parameters.
Talk Properties: Output Settings
Table 3-128 describes the Talk function Output parameters.
Table 3-126. Talk - Parameters
Name Description
Delay Before Execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Playback Settings When the Overwrite playback activity setting option is selected, the following configured playback settings overrride the activity-level settings:
• Clip: Selects a wave file or a QoV clip or to be played
back by the script function. Clicking the : opens a window that permits you to select the audio file from the IxLoad Resource Manager.
• Output level: In the case of QoV clips, this specifies an output level of -20, -25, -30, or -35 dBm.
• Play: Specifies the number of times to play.
• Repeat continuously for: Specifies a period of time to play. The default value is 1000 seconds.
• Use Talk Time (for all objectives except Channels): Plays the wave file for the duration of the Talk Time call parameter.
• Use Global settings: The playback values are those specified in the Global Settings | T1/E1 | Voice win-dow.
Table 3-127. Talk - Advanced Settings Parameters
Name Description
Stop playback on first detected DTMF
If selected, the clip playback is stopped when a specified DTMF is detected.
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Listen This script function allows the recording of a wave file or QoV clip for a speci-fied duration.
Listen Properties: Parameters
Table 3-129 describes the Listen function parameters.
Table 3-128. Talk - Output Settings
Properties Description
Output Settings
Ok The function was executed correctly executed for the specified number of times or duration.
The default resolution for this output is SUCCESS.
Error The function has returned an error.
The default resolution for this output is FAILED.
Table 3-129. Listen - Parameters
Name Description
Delay Before Execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Listen Settings Specifies the following recording settings:
• Data Format: The data format can be Mu-Law, A-Law, PCM, or as specified by the Global Settings | T1/E1 | Voice window. Default it is Global.
• Sampling Rate: The sampling rate value can be 6000 Hz, 8000 Hz, 11025 Hz, or as specified by the Global Settings | T1/E1 | Voice window.
• Bits per sample: The sample size can be 8 bit, 16 bit, or as specified by the Global Settings | T1/E1 | Voice window.
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Listen Properties: Advanced Settings
Table 3-130 describes the Voice Session function Advanced Playback parame-ters.
Listen Properties: Output Settings
Table 3-131 describes the Listen function Output Settings parameters.
Voice Session This script function plays back a wave file or QoV clip and records an audio file at the same time.
When the Overwrite playback activity setting option is selected, the following configured playback settings overrride the activity-level settings:
• Listen Duration: If selected, the recording duration specified as a value or a formula (expression). The maximum duration value is 10 minutes. By default this option is selected and configured to a 10 s value.
• Use Talk Time (for all objectives except Channels): If selected, the recording duration is set to the value of the Talk Time call parameter.
Perform QoV measurements
If selected, P.862 PESQ and P56 QoV scores computation is performed for the reference audio file specified in the Clip field. The clip’s volume is specified by the adjacent Output level field.
Table 3-130. Listen - Advanced Settings Parameters
Name Description
Listen Terminate Conditions - Stop recording on first detected DTMF
If selected, the clip recording is stopped when a specified DTMF is detected.
Table 3-131. Talk - Output Settings
Properties Description
Output Settings
Ok The function was executed correctly executed for the specified number of times or duration.
The default resolution for this output is SUCCESS.
Error The function has returned an error.
The default resolution for this output is FAILED.
Table 3-129. Listen - Parameters (Continued)
Name Description
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Voice Session Properties: Talk Parameters
The Voice Session Talk parameters are the same as those from Talk Properties: Parameters on page 3-163.
Voice Session Properties: Listen Parameters
The Voice Session Talk parameters are the same as those from Listen Properties: Parameters on page 3-165.
Voice Session Properties: Advanced Settings
Table 3-132 describes the Voice Session function Advanced Playback parame-ters.
Voice Session Properties: Output Settings
Table 3-133 describes the Voice Session function Output Settings parameters.
Generate DTMF This script function generates a sequence of Dual Tone Multiple Frequency (DTMF) signals. The sequence of tones can have up to 31 DTMFs and the tone length and the inter-tone interval can be specified. The sequence may contain any combination of standard tones (that is, "1", "2", "3", "4", "5" "6", "7", "8", "9" "0", "#", "*", "a" "b", "c", "d").
Generate DTMF Properties: Parameters
Table 3-134 describes the Generate DTMF function parameters.
Table 3-132. Voice Session - Advanced Settings Parameters
Name Description
Stop playback on first detected DTMF
If selected, the audio file playback is stopped when a specified DTMF is detected.
Stop recording on first detected DTMF
If selected, the audio file recording is stopped when a specified DTMF is detected.
Table 3-133. Voice Session - Output Settings
Properties Description
Output Settings
Ok The function was executed correctly executed for the specified number of times or duration.
The default resolution for this output is SUCCESS.
Error The function has returned an error.
The default resolution for this output is FAILED.
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Generate DTMF Properties: Advanced Settings
Table 3-135 describes the Generate DTMF Advanced Settings parameters.
Table 3-134. Generate DTMF - Parameters
Name Description
Delay Before Execution
Delays the function execution by a value specified as either of the following:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Tone duration The time length of a DTMF tone in the tone sequence. The range of values is Global, or 50 to 10000 ms.
The default value is 200 ms.
InterDigit Delay The delay between 2 consecutive DTMF tones in the tone sequence, including the silence time. The range is Global, or 50 to 10000 ms.
The default value is 200 ms.
Signal power The amplitude of the generated DTMF tones, measured in decibels (dBm). The range of values can be Global ,or from -54 to -3 dBm.
The default value is -3 dBm.
DTMF Sequence The sequence of DTMF tones to be generated, which can be local or can be chosen from the DTMF Pool. The default is the ‘12345’ DTMF sequence.
Table 3-135. Generate DTMF - Advanced Settings
Name Description
Playback Specifies the DTMF playback mode using either of the options below, or by applying the values specified in the Global Settings | T1/E1 | Voice window (for the Use Global Setting option):
• Play: Specifies the number of times to play.
• Repeat Continuously for: Specify the period of time to play. The default value is 1000 seconds.
• Use Talk Time (All objectives except Channels): Plays the DTMF sequence for the duration of the Talk Time call parameter.
• Use Global Settings: If selected, the global settings configured in the Global Settings | T1/E1 | Voice page are used.
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Generate DTMF Properties: Output Settings
Table 3-136 describes the Generate DTMF function Output Settings parameters.
Detect DTMF This script function is used to detect a sequence of DTMF signals. The sequence can have up to 31 tones and the inter-tone interval can be specified. The inter-tone interval includes the silence time between two consecutive tones. The expected sequence to be received may contain any of the standard tones (that is, '1', '2', '3', '4', '5', '6', '7', '8', '9', '0', '#', '*', 'a', 'b', 'c', 'd').
Detect DTMF Properties: Parameters
Table 3-137 describes the Detect DTMF function parameters.
Table 3-136. Voice Session - Output Settings
Properties Description
Output Settings
Ok The function was executed correctly executed for the specified number of times or duration.
The default resolution for this output is SUCCESS.
Error The function has returned an error.
The default resolution for this output is FAILED.
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Detect DTMF Properties: Output Settings
Table 3-138 describes the Detect DTMF Output Settings parameters.
Table 3-137. Detect DTMF - Parameters
Name Description
Delay Before Execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
DTMF Detection Settings
Specifies the following detection parameters:
• Detect Continuously for: Detects all the DTMFs arrived in the specified period of time (measured in seconds, minutes or hours). It can be a value or a for-mula (expression). Maximum duration value is 10 minutes. The default value is 1000 seconds.
• Detect Exactly: Detects the specified number of DTMFs. The default value is 6 DTMFs.
• Use Talk Time (all objectives except Channels): Per-forms detection for the duration of the Talk Time call parameter.
• Detect DTMF Sequence: Detects the expected sequence (specified by selecting an entry in the DTMF Sequence pool items, or a Local Sequence). The default is 12345 DTMF string.
Terminate Conditions
Specifies the termination condition of the DTMF detection as either of the following:
• Maximum delay between tones: If selected, the DTMF detection is terminated when tones arrive further apart than the specified value. The value is Global, or ranging from 50 to 30000 ms.
• First tone timeout: If selected, DTMF detection is ter-minated when the first tone arrives later that the spec-ified amount of time. The value is Global, or ranging from 50 to 10000 ms.
Threshold Value: If selected, this parameter configures a threshold value for DTMF detection, whereby DTMFs having the duration smaller than the threshold value are not detected. The value range is 50 to 10000 ms.
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Generate MF This script function generates a sequence of Multi - Frequency (MF) tones. The sequence can have up to 31 tones and the tone length and the inter-tone Interval can be specified. The sequence can contain any combination of the standard tones, (that is, '1', '2', '3', '4', '5', '6', '7', '8', '9', '0', '#', '*', 'a', 'b', 'c').
Since its parameters are similar to these of the Generate DTMF script function, see Generate DTMF on page 3-167 for detailed information.
Detect MF This script function is used to detect a sequence of MF tones. The sequence can have up to 31 tones and the inter-tone interval can be specified. Note that the inter-tone interval includes the silence time between 2 consecutive tones. The expected sequence to be received can contain any of the standard tones (that is, '1', '2', '3', '4', '5', '6', '7', '8', '9', '0', '#', '*', 'a', 'b', 'c').
Since its parameters are similar to these of the Detect DTMF script function, see Detect DTMF on page 3-169 for detailed information.
Generate Tone This script function generates a single custom tone (single or dual continuous, or cadence) that can be selected from the Custom Tones Pool. The tones frequency must be between 300 and 3500 Hz, and, for dual tones, the difference of fre-quency between the 2 tones must be at least 65 Hz.
Generate Tone Properties: Parameters
Table 3-139 describes the Generate Tone function parameters.
Table 3-138. Detect DTMF - Output Settings
Properties Description
Connected The DTMF detection was correctly executed for the specified number of times or duration.
The default resolution for this output is SUCCESS.
Timeout The DTMF detection failed due to a timeout (e.g. DTMF/MF/Tone detection timeout).
The default resolution for this output is WARNING.
Error The function has returned an error.
The default resolution for this output is FAILED.
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Generate Tone Properties: Advanced Settings
Table 3-140 describes the Generate Tone Advanced Settings parameters.
GenerateTone Properties: Output Settings
Table 3-141 describes the Generate Tone function Output Settings parameters.
Table 3-139. Generate Tone- Parameters
Name Description
Delay Before Execution
Delays the function execution by a value specified as either of the following:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Tone Name Specifies a custom tone
Tone duration The time length of a DTMF tone in the tone sequence. The value is the global value configured in Global Settings | T1/E1 | Voice window (for the Global option), or ranging from 50 to 10000 ms.
The default value is 200 ms.
Table 3-140. Generate Tone - Advanced Settings
Name Description
Playback Specifies the tone playback mode using either of the options below, or by applying the values specified in the Global Settings | T1/E1| Voice window (for the Use Global Setting option):
• Play: Specifies the number of times to play.
• Repeat Continuously for: Specify the period of time to play. The default value is 1000 seconds.
• Use Talk Time (All objectives except Channels): Plays the tone for the duration of the Talk Time call parame-ter.
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Wait for Tone This script function detects any custom tone from a user-configured tone list. For each expected tone that is configured in the list, a corresponding output is added to the script function; when a particular tone from the list is detected, the function exits on the corresponding output.
Wait for Tone Properties: Options
Table 3-142 describes the Wait for Tone function parameters.
Table 3-141. Generate Tone - Output Settings
Properties Description
Output Settings
Ok The function was executed correctly executed for the specified number of times or duration.
The default resolution for this output is SUCCESS.
Error The function has returned an error.
The default resolution for this output is FAILED.
Table 3-142. Wait for Tone - Parameters
Name Description
Delay Before Execution
Delays the function execution by a value specified as:
• Static Expression, in milliseconds (ms)
• Random Between Expressions, in milliseconds (ms)
• Sleep 1000, in milliseconds (ms)
• GetCallInfo Delay, in milliseconds (ms)
• Detect DTMF delay, in milliseconds (ms)
• Generate DTMF delay, in milliseconds (ms)
• Sleep 2000, in milliseconds (ms)
Tone Detection Settings
This area contains a list of tones to be detected.
Initially the list is empty, adding a tone is done by clicking
the , which displays a dialog that allows you to specify the expected tone.
An tone can be deleted from the tone list by selecting its
entry and clicking the button.
Tone timeout: Specifies a timeout by entering a custom value, or by applying the global timeout settings configured in the Global Settings | T1/E1 | Voice window (for the Global value ).
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Wait for Tone Properties: Output Settings
Table 3-125 describes the Wait for Tone Output Settings parameters. As stated before, a new script function output with the same name as the expected custom tone is added.
Table 3-143. Wait for Tone - Output Settings
Properties Description
Output Settings
Custom Tone #1
The specified custom tone (Custom Tone #1) was detected.
The default resolution for this output is SUCCESS.
Custom Tone #2
The specified custom tone (Custom Tone #2) was detected.
The default resolution for this output is SUCCESS.
Timeout The detection operation failed due to a timeout.
The default resolution for this output is WARNING.
Error The function has returned an error.
The default resolution for this output is FAILED.
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4Chapter 4: Basic Test Scenarios and Procedures
This chapter describes the built-in procedures and sample test configurations pro-vided with the IxLoad Voice Plug-in and covers the following topics:
• SIP Procedures, Sample Test Configurations and Test Scenarios on page 4-1.
• Skinny Procedures, Sample Test Configurations and Test Scenarios on page 4-84
• H.323 Sample Test Configurations and Test Scenarios on page 4-169
• H.248 Sample Test Configurations and Test Scenarios on page 4-181
• MGCP Sample Test Configurations and Test Scenarios on page 4-210
• PSTN Sample Test Configurations and Test Scenarios on page 4-228
SIP Procedures, Sample Test Configurations and Test Scenarios
Using the functions from the VoIP SIP test library, you can generate and execute a large number of originator/answerer scenario configurations that comply with the SIP protocol. You can use one of the predefined test scenarios described in this chapter or create a new one, map it to VoIPSIP Peer activities, and start the execution.
This section describes the pre-defined IxLoad Voice Plug-in SIP procedures, available sample test configurations (RXFs) and their associated test scenarios.
Note: For a complete description of the supported SIP test library functions, refer to the VoIP SIP Functions Library on page 3-1.
Note: Sample tests follow a naming convention that comprises test type (VS for VoIP SIP and SC for SIP Cloud), an index, a test topology (T1 shown in Figure 5-22 or T2 shown in Figure 5-23), a configuration (B2B for tests in back-to-back mode or DUT for tests running against a DUT), a protocol version (SIPv4 or SIPv6) and a short call description, such as for example SC_001_B2B_SIPv4_T1_MakeCall_from_Cloud.
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SIP Predefined Procedures
A procedure is a simple way to encapsulate several script functions into a single function block that can be re-used within a number of scenarios.
Based on the functions from the SIP test library, the SIP predefined procedures described in Table 4-1 (available in the Procedure Library) were developed.
Table 4-1. SIP Low-Level Predefined Procedures
Category Procedure Description
Registration SIP MakeRegistration A registration procedure that resolves 1xx and 200 response messages.
SIP Make Registration - Authentication
A registration procedure resolving 1xx, 200, and 401 response messages. The procedure is used in test scenarios that require authentication.
SIP Make Registration - 407 - Authentication
A registration procedure similar to the previous one, except that it additionally resolves 407 response messages.
SIP MakeRegistration - First Loop Only
A registration procedure similar to the previous MakeRegistration - Authentication procedure, except that the SIP endpoint registers only during the first loop execution.
SIP MakeRegistration 407 - First Loop Only
A registration procedure similar to the previous one, executing registration during the first loop only. The procedure additionally resolves 407 response messages.
SIP Make Registration - Complete
A registration procedure that performs registration for the first loop only and resolves 1xx, 200, 401 and 5xx responses. When retransmissions are required, they need to be enabled at activity level.
SIP Make ReRegistration - Authentication
A registration procedure that performs an initial registration during the first loop and retrieves the value of the expires SIP parameter or header.
Starting with the second loop, the procedure evaluates the expiration timer and performs re-registration if the current registration is to expire during the current loop.
SIP Unregister A de-registration procedure that unregisters only one binding associated with an address-of-record.
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SIP Unregister All Bindings
A de-registration procedure that removes all the previous bindings currently in place for an address-of-record.
Originate Call
SIP MakeCall A call originating procedure that comprises the Send INVITE, Wait 1xx, Wait 200, Send ACK functions sequence.
SIP Make Call - Authentication
A call originating procedure that comprises the Send INVITE, Wait 1xx, Wait 200, or Wait 407, Send ACK, Re-send INVITE with Authentication, Wait 1xx or Wait 200, Send ACK functions sequence.
SIP Make Call - Route
A call originating procedure that uses routing information.
This is the same as the previous Make Call - Authentication procedure, except that the Send ACK script function includes the SIP Route message header.
This procedure is used in test scenarios that require routing functionality.
SIP Make Call - Redirect Server
A call originating procedure that resolves 3xx response messages and uses redirect information. This is the same as the previous Make Call - Authentication procedure, except that it is used in scenarios that involve a redirect server.
SIP Make Call - Complete
A call originating procedure that resolves a wide range of responses: 1xx, 200, 3xx, 4xx, 5xx, and 6xx.
Note: In order to use the message retransmission mechanism, this procedure needs to have retransmissions enabled at activity level.
Receive Call SIP Receive Call A call receiving procedure that comprises the Wait INVITE, Send 180 Ringing, Send 200, Wait ACK functions sequence.
SIP Receive Call - Busy Here
A call receiving procedure that sends a 486 Busy here message in response to a SIP INVITE message.
Table 4-1. SIP Low-Level Predefined Procedures (Continued)
Category Procedure Description
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SIP Receive Call - No Answer
A call receiving procedure that sends a 180 Ringing message in response to a SIP INVITE message and then waits for a SIP CANCEL message.
SIP Receive Call - Record Route
A call receiving procedure similar to the previous Receive Call procedure, except that SIP response messages include the Record-Route message header field.
This procedure is used in test scenarios that require routing functionality.
End Call SIP End Call Initiate A simple call disconnection procedure that comprises a Send BYE and Wait 200 functions sequence.
SIP End Call Initiate - Route
A call disconnection procedure similar to the previous End Call Initiate procedure, except that it uses routing information.
This procedure is used in test scenarios that require routing functionality.
SIP End Call Receive A call disconnection procedure for the receiving side, which comprises a Wait BYE and Send 200 OK functions sequence.
SIP End Call Receive - Record Route
A call disconnection procedure similar to the previous procedure, except that the Send 200 OK function includes a Record-Route header.
This procedure is used in scenarios that require routing functionality.
Hold/Unhold SIP Hold - Initiate A procedure that puts the remote party on hold and comprises a Send INVITE (Hold Session message body), Wait 200 OK, Send ACK functions sequence.
SIP UnHold - Initiate A procedure that unholds the remote party and comprises a Send INVITE, Wait 200 OK, Send ACK commands sequence.
SIP Hold - Receive A procedure that puts the party on hold. It comprises a Wait INVITE, Send 200 OK, Wait ACK commands sequence.
SIP UnHold - Receive A procedure that takes the party off hold. The procedure comprises a Wait INVITE, Send 200 OK, Wait ACK commands sequence.
Table 4-1. SIP Low-Level Predefined Procedures (Continued)
Category Procedure Description
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IMS SIP IMS MakeRegistration
An IMS-compliant registration procedure that uses custom message headers and user-defined scenario variables.
Note: The scenario variables used by the procedure are described in the procedure body.
SIP IMS MakeCall An IMS calling procedure for the originating side that comprises the Send INVITE, Wait 100 Trying or 183 Session in Progress, Send PRACK, Wait 200 OK for PRACK, Send UPDATE, Wait 200 OK for UPDATE, Wait 180 Ringing, Send PRACK, Wait 200 OK for PRACK, Wait 200 OK for INVITE, Send ACK functions sequence.
SIP IMS ReceiveCall An IMS procedure for the receiving side, comprising the Wait INVITE, Send 100 Trying or 183 Session in Progress, Wait PRACK, Send 200 OK for PRACK, Wait UPDATE, Send 200 OK for UPDATE, Send 180 Ringing, Wait PRACK, Send 200 OK for PRACK, Send 200 OK for INVITE, Wait ACK messages sequence.
SIP IMS EndCall Initiate
An IMS call terminating procedure for the originating side comprising a Send BYE, Wait 200 OK functions sequence.
SIP IMS EndCall Receive
An IMS call terminating procedure for the receiving side comprising a Wait BYE, Send 200 OK functions sequence.
SIP IMS Subscribe An IMS subscription procedure that comprises the Send SUBSCRIBE, Wait 200 OK, Wait NOTIFY, Send 200 OK for NOTIFY (Wait multiple NOTIFY) functions sequence.
Instant Messaging Support
SIP Send Instant Message
An instant message sending procedure that comprises the Send MESSAGE and Wait 200 OK functions sequence.
SIP Wait Instant Message
An instant message receiving procedure that comprises the Wait MESSAGE and Send 200 OK functions sequence.
SMS SIP SMS Deliver Initiate
SIP SMS Deliver Initiate is used to simulate CSCF behavior for sending Deliver messages
The procedure comprises Send Deliver, Wait 200OK, Wait Deliver-Report, and Send 202 Accepted functions set.
Table 4-1. SIP Low-Level Predefined Procedures (Continued)
Category Procedure Description
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The following section provides you with a short description of the SIP test sce-narios that are included in the IxLoad installation kit.
VoIPSIP Peer Test Configurations
This section provides a brief description of the sample VoIPSIPPeer test configu-ration files contained in the IxLoad installation kit.
SIP SMS Deliver Receive
SIP SMS Deliver Receive is used to simulate Endpoint behavior for receiving Deliver messages.
The procedure comprises Wait Deliver, Send 200 OK, Send Deliver-Report and Wait 202 Accepted functions set.
SIP SMS Status-Report Initiate
SIP SMS Status-Report Receive is used to simulate CSCF behavior for sending Status-Report messages.
The procedure comprises Send Status-Report, and Wait 200 OK functions set.
SIP SMS Status-Report Receive
SIP SMS Status-Report Receive is used to simulate Endpoint behavior for receiving Status-Report messages.
The procedure comprises Wait Status-Report, and Send 200 OK functions set.
SIP SMS Submit Initiate
SIP SMS Submit Initiate is used to simulate Endpoint behavior for sending Submit messages.
The procedure comprises Send Submit, Wait 202 Accepted, Wait Submit-Report, and Send 200 OK functions set.
SIP SMS Submit Receive
SIP SMS Submit Receive is used to simulate CSCF behavior for receiving Submit messages
The procedure comprises Wait Submit, Send 202 Accepted, Send Submit-Report and Wait 200 OK functions set.
Other First Loop? A procedure that tests if the current loop is the first loop.
Table 4-1. SIP Low-Level Predefined Procedures (Continued)
Category Procedure Description
Note: Sample SIP test configurations do not have retransmissions enabled at activity level. For complex tests that require retransmissions you can also enable retransmissions at function level.
Note: When configuring SIP UAs, keep in mind that a signaling endpoint has to be configured using a unique (IP address, Port, Phone number) tuple, while a media endpoint is uniquely identified by a (IP Address, Port) tuple.
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VS_001_B2B_SIPv4 MakeCall - ReceiveCall - EndCall
This test is similar to the following VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s test, except that in the underlying test scenario Sleep script function are used instead of the Voice Session functions.
VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s
This test runs in Back-to-Back mode and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that establishes a SIP call with media streaming using Voice Session script functions and then disconnects, as shown in Figure 4-1. Receive_Call executes the test flow on the receiving side.
Note: Sample tests provided with the IxLoad Voice Plug-in normally do not resolve 407 response messages. In cases when your tests require the resolving of 407 messages, predefined registration procedures from the sample scenarios can be replaced with the SIP Make Registration - 407 - Authentication and the SIP Make Registration - 407 - First Loop Only procedures that handle such messages.
Note: The SIP sample templates include a VS_034_B2B_SIPv6 MakeCall - ReceiveCall - EndCall predefined test identical to this one, except that it uses IPv6 network settings.
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Figure 4-1. VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s Test Scenario
The Make_Call configuration settings are described in Table 4-2.
Table 4-2. Make_Call Activity Test Settings
Category Settings
Scenario The VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s test scenario comprises 2 channels executing a basic call procedure with media exchange.
Channel#0: MakeCall, Voice Session, EndCall Initiate.
Channel#1: ReceiveCall, Voice Session, EndCall Terminate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
Dial Plan The Receive_Call activity and port 5060 are configured as call destination.
SIP Settings The SIP port has the 5060 default setting for all channels.
Codec Settings The default codec settings are used.
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VS_003_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with SRTP - 33s
This test is similar to the previous VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s test, except that the media traffic is encrypted using SRTP.
VS_004_B2B_SIPv4 MakeCall - ReceiveCall - EndCall - Early Media
This test is similar to the previous VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s test, except that the media capabilities are negotiated on a provisional 183 SessionInProgress response instead of a 200 OK response.
RTP Settings The Enable media on this activity option is selected for media streaming to be performed.
Since Make_Call emulates media endpoints using different IP addresses, a single RTP port (10000) is configured for media streaming.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
Note: The SIP sample templates include a VS_035_B2B_SIPv6 MakeCall - ReceiveCall - EndCall with RTP - 33s predefined test identical to this one, except that it uses IPv6 network settings.
Note: The SIP sample templates include a VS_036_B2B_SIPv6 MakeCall - ReceiveCall - EndCall with SRTP - 33s predefined test identical to this one, except that it uses IPv6 network settings.
Table 4-2. Make_Call Activity Test Settings (Continued)
Category Settings
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Figure 4-2. VS_004_B2B_SIPv4 MakeCall - ReceiveCall - EndCall - Early Media
The Make_Call configuration settings are described in Table 4-3.
Table 4-3. Make_Call Activity Test Settings
Category Settings
Scenario The VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s test scenario comprises 2 channels executing a basic call procedure with media exchange.
Channel#0: MakeCall, Voice Session, EndCall Initiate.
Channel#1: ReceiveCall, Voice Session, EndCall Terminate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
Dial Plan The Receive_Call activity and port 5060 are configured as call destination.
SIP Settings The SIP port has the 5060 default setting for all channels.
Codec Settings The default codec settings are used.
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VS_005_DUT_SIPv4 MakeCall - ReceiveCall - EndCall with RecordRoute
This test running against a SIP server comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that registers with a SIP server and then establishes a SIP call without media streaming, remains idle for the config-ured duration of the Sleep script function, and then disconnects, as shown in Figure 4-3.
A Receive_Call -emulated endpoint registers with a SIP server and then executes the test flow on the receiving side.
Figure 4-3. VS_005_DUT_SIPv4 MakeCall - ReceiveCall - EndCall with RecordRoute Test Scenario
RTP Settings The Enable media on this activity option is selected for media streaming to be performed.
Since Make_Call emulates media endpoints using different IP addresses, a single RTP port (10000) is configured for media streaming.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
Note: The SIP sample templates include a VS_035_B2B_SIPv6 MakeCall - ReceiveCall - EndCall with RTP - 33s predefined test identical to this one, except that it uses IPv6 network settings.
Table 4-3. Make_Call Activity Test Settings (Continued)
Category Settings
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The Make_Call configuration settings are described in Table 4-4.
VS_006_DUT_SIPv4 MakeCall - ReceiveCall with Registration
This test runs against a SIP Server and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that registers with a SIP Proxy server, establishes a SIP call without media traffic, and disconnects, as shown in Figure 4-4. Receive_Call executes the test flow for the receiving side.
Table 4-4. Make_Call Activity Test Settings
Category Settings
Scenario The VS_005_DUT_SIPv4 MakeCall - ReceiveCall - EndCall with RecordRoute test scenario is completely configured.
The SIP MakeCall - Route and SIP EndCall Initiate - Route procedures used on the first channel are used for configuring message routing functionality.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity and port 5060 are configured as call destination.
SIP Settings The SIP port has the 5060 default setting for all channels.
The SIP server the test runs against is configured in the Use external server area.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test does not perform any media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
Note: The SIP sample templates include another VS_039_DUT_SIPv6 MakeCall - ReceiveCall - EndCall with RecordRoute predefined test identical to this one, except that it uses IPv6 network settings.
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Figure 4-4. VS_006_DUT_SIPv4 MakeCall - ReceiveCall Call with Registration Test Scenario
Make_Call configuration settings are described in Table 4-5.
Table 4-5. Make_Call Activity Test Settings
Category Settings
Scenario Editor The VS_006_DUT_SIPv4 Make - Receive Call with Registration test scenario is completely configured and supports authentication at test scenario level.
The scenario comprises 2 channels:
Channel#0: Make Registration, MakeCall - Authentication, EndCall Initiate.
Channel#1: Make Registration, ReceiveCall, EndCall Receive.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as call destination.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The SIP server you are running this test against must be configured in the Use external Server area.
Note: Since the test scenario itself supports authentication, you can configure the test to use authentication and enter the desired authentication settings in the Authentication UAC area.
Codec Settings The codec settings can be left unaltered.
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VS_007_DUT_SIPv4 Make - Receive Call with ReRegistration
This test runs against a SIP Server and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call emulates an endpoint that registers with a SIP server using the SIP Make ReRegistration - Authentication procedure, establishes a signaling-only call with another endpoint emulated by Receive_Call, and disconnects, as shown in Figure 4-5. Receive_Call executes the test flow for the receiving side.
Figure 4-5. VS_007_DUT_SIPv4 Make-Receive Call with ReRegistration Test Scenario
The Make_Call configuration settings are described in Table 4-6.
RTP Settings Since this test does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
Note: The SIP sample templates include a VS_037_DUT_SIPv6 MakeCall - ReceiveCall with Registration predefined test identical to this one, except that it uses IPv6 network settings.
Table 4-5. Make_Call Activity Test Settings
Category Settings
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VS_008_DUT_SIPv4 MakeCall - Receive Call with Registration - Complete
This test runs against a SIP Server and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Table 4-6. Make_Call Activity Test Settings
Category Settings
Scenario Editor The VS_007_DUT_SIPv4 Make - Receive Call with ReRegistration test scenario is completely configured and supports authentication at scenario level.
The scenario comprises 2 channels.
Channel#0: Make ReRegistration - Authentication, MakeCall - Authentication, Sleep, EndCall Initiate.
Channel#1: Make ReRegistration - Authentication, ReceiveCall, Sleep, EndCall Receive.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The SIP server you are running this test against needs specified in the Use external server area.
Note: Since the test scenario itself supports authentication, you can configure the test to use authentication and enter the desired authentication settings in the UAC Authentication area.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
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Make_Call is linked to a test scenario channel that emulates an endpoint that reg-isters with a SIP server, establishes a SIP call with an endpoint emulated by Receive_Call, and then disconnects, as shown in Figure 4-6. Receive_Call exe-cutes the test flow on the receiving side.
Figure 4-6. VS_008_DUT_SIPv4 MakeCall - ReceiveCall with Registration - Complete Test Scenario
VS_009_B2B_SIPv4 MakeCall - ReceiveCall with Tel URI - Global Phone Numbers
This test is identical to the following VS_010_B2B_SIPv4 Make - Receive Call with Tel URI - Local Phone Numbers test, except that it uses a global tel URI des-tination in the Dial Plan page. When a global tel URI is used, no additional parameters need to be specified.
VS_010_B2B_SIPv4 Make - Receive Call with Tel URI - Local Phone Numbers
This test runs in Back-to-Back mode and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that establishes a call using the Make Call - Authentication procedure, and disconnects using the End Call Initiate procedure, as shown in Figure 4-7. Receive_Call executes the corre-sponding operations on the receiving side.
Note: The SIP sample templates include a VS_038_DUT_SIPv6 MakeCall - ReceiveCall with Registration - Complete predefined test identical to this one, except that it uses IPv6 network settings.
Note: The underlying test scenario is the same for both this and the following VS_010_B2B_SIPv4 Make - Receive Call with Tel URI - Local Phone Numbers test configuration.
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Figure 4-7. VS_010_B2B_SIPv4 MakeCall - ReceiveCall with Tel URI Test Scenario
The Make_Call settings are described in Table 4-7.
Table 4-7. Make_Call Activity Test Settings
Category Settings
Scenario Editor The VS_010_B2B_SIPv4 MakeCall - ReceiveCall with Tel URI test scenario is completely configured, no further configuration is necessary.
Execution Settings The channels are configured to use consecutive IPs from those available in the Network settings.
For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as call destination.
Since the test uses local Tel URIs, the phone-context parameter according to RFC 3966 must be defined for the phone number sources.
SIP Settings The SIP port has the 5060 default setting.
In the Construction of SIP messages area, the Use Tel URI scheme for source and Use Tel URI scheme for destination options are selected.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test does not use RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4.
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The Receive_Call settings are described in Table 4-8.
Table 4-8. Receive_Call Activity Test Settings
Category Settings
Scenario Editor The VS_010_B2B_SIPv4 MakeCall - ReceiveCall with Tel URI test scenario is completely configured, no further configuration is necessary.
Execution Settings The channels are configured to use consecutive IPs from those available in the Network settings.
For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan For this activity which only terminates a call, no VoIPSIPPeer activity is configured as destination activity.
Source phone numbers are defined as a sequence generator expression. Since the test uses local Tel URIs, the phone-context parameter according to RFC 3966 must be defined for the phone number sources.
SIP Settings The SIP port has the 5060 default setting.
In the Construction of SIP messages area, the Use Tel URI scheme for source is selected.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test does not use RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4.
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VS_011_B2B_SIPv4 Basic Transfer - Successful
This test comprises three VoIPSIPPeer activities, VoIPSIPPeer1, VoIPSIPPeer2, and VoIPSIPPeer3.
VoIPSIPPeer2 (transferee) is linked to a test scenario channel that establishes a SIP call with VoIPSIPPeer1 (transferrer), then transfers the call to VoIPSIPPeer3 (transfer target), as shown in Figure 4-8.
Figure 4-8. VS_011_B2B_SIPv4 Basic Transfer - Successful Test Scenario
The VoIPSIPPeer1 configuration settings are described in Table 4-9.
Table 4-9. VoIPSIPPeer1 Activity Test Settings
Category Settings
Scenario The VS_011_B2B_SIPv4 Basic Transfer - Successful test scenario comprises 3 channels that implement a call transfer:
• Channel#0: This is the receiving party for the call initi-ated by endpoint B. Endpoint A (the transferrer) puts the established call on hold and transfers it to C.
• Channel#1: This is the originating party for a call to endpoint A. After the call is established, endpoint B (the transferee) is transferred to endpoint C.
• Channel#2: Endpoint C is the transfer target.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan Since the activity only terminates a call, no activity is configured as call destination.
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VS_012_B2B_SIPv4 Basic Transfer - Target Busy
The test configuration level settings are the same as those for the previous VS_011_B2B_SIP v4 Basic Transfer - Successful test.
VS_013_B2B_SIPv4 Basic Transfer - Target No Answer
The test configuration level settings are the same as those for the previous VS_011_B2B_SIPv4 Basic Transfer - Successful test.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Important: The address of the VoIPSIP Peer activity to which the call is transferred (VoIPSIPPeer3) is configured as Transfer Address in the SIP Settings page.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: VoIPSIPPeer2 uses the same settings as VoIPSIPPeer1, except for the Dial Plan page, which specifies VoIPSIPPeer1 as call destination.
The settings for VoIPSIPPeer3 are similar to those of VoIPSIPPeer1, except for the transfer address, which needs not be specified in the SIP Settings page.
Note: The SIP sample templates include another VS_041_B2B_SIPv6 Basic Transfer - Successful predefined test identical to this one, except that it uses IPv6 network settings.
Note: The only difference between the VS_012_B2B_SIPv4 Basic Transfer - Target Busy and the VS_011_B2B_SIPv4 Basic Transfer - Successful tests is the underlying scenario, which simulates a transfer failure due to a busy target condition, instead of a successful transfer.
Note: The only difference between the VS_013_B2B_SIPv4 Basic Transfer - Target No Answer and the VS_011_B2B_SIPv4 Basic Transfer - Successful tests is the underlying scenario, which simulates a transfer failure due to a target no answer condition, instead of a successful transfer.
Table 4-9. VoIPSIPPeer1 Activity Test Settings
Category Settings
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VS_014_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with Hold UnHold
This test illustrating a Hold / Unhold procedure runs in Back-to-Back mode and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that establishes a SIP call without media streaming, performs a hold/unhold procedure using the SIP Hold and SIP Unhold script functions, and then disconnects, as shown in Figure 4-9. Receive_Call executes the test flow on the receiving side.
Figure 4-9. VS_014_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with Hold Unhold Test Scenario
The Make_Call configuration settings are described in Table 4-10.
Table 4-10. Make_Call Activity Test Settings
Category Settings
Scenario The VS_014_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with Hold - Unhold test scenario comprises 2 channels:
Channel#0: MakeCall, SIP Hold - Initiate, Sleep, Unhold - Initiate, EndCall Initiate.
Channel#1: ReceiveCall, Hold - Receive, Sleep, Unhold - Receive, EndCall Terminate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity and port 5060 are configured as call destination.
SIP Settings The SIP port has the 5060 default setting for all channels.
Codec Settings The codec settings can be left unaltered.
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VS_015_DUT_SIPv4 Hold - UnHold with Registration and Path Confirmation
This test runs against a SIP server and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that registers with a SIP server, establishes a call, performs audio path confirmation using the Generate DTMF/ Detect DTMF script functions pair, puts the remote party on hold, unholds the remote party, performs audio path confirmation again, and disconnects, as shown in Figure 4-10 on page 4-22. Receive_Call executes the corresponding test flow on the receiving side.
Figure 4-10. VS_015_DUT_SIPv4 Hold-Unhold with Registration and Path Confirmation Test Scenario
The Make_Call configuration settings are described in Table 4-11.
RTP Settings Since this test does not perform any media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
Note: The SIP sample templates include another VS_040_B2B_SIPv6 MakeCall - ReceiveCall - EndCall with Hold Unhold predefined test identical to this one, except that it uses IPv6 network settings.
Table 4-10. Make_Call Activity Test Settings (Continued)
Category Settings
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Table 4-11. Make_Call Activity Test Settings
Category Settings
Scenario The VS_015_DUT_SIPv4 Hold - UnHold with Registration, Path Confirmation test scenario is completely configured and supports authentication at test scenario level.
Channel#0: Make Registration, Make Call, Generate DTMF, Detect DTMF, Hold, Sleep, UnHold, Generate DTMF, Detect DTMF, End Call Initiate.
Channel#1: Make Registration, Receive Call, Detect DTMF, Generate DTMF, onHold, Sleep, onUnHold, Detect DTMF, Generate DTMF, End Call Receive.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call activity is configured as call destination.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port: Use same value setting from the Execution Settings page.
The SIP server you are running this test against must be configured in the Use external server area.
Note: Since the test scenario itself supports authentication, you can configure the test to use authentication and enter the desired authentication settings in the UAC Authentication area.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test uses RTP streaming on the Generate DTMF/Detect DTMF script functions, the Enable media on this activity option is selected.
For this test, which uses consecutive values for the IP addresses, RTP ports for all channels can be configured to the default 10000 setting.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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VS_016_DUT_SIPv4 Send - Receive MESSAGE with Registration
This test comprises two VoIPSIPPeer activities, Make_Call and Receive_Call, whose emulated endpoints register with a SIP Proxy server using SIP MakeReg-istration - First Loop Only procedures and then exchange a SIP MESSAGE message, as shown in Figure 4-11.
Figure 4-11. VS_016_DUT_SIPv4 Send - Receive MESSAGE with Registration
The Make_Call configuration settings are described in Table 4-12.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
Table 4-12. Make_Call Activity Test Settings
Category Settings
Scenario Editor The VS_016_DUT_SIPv4 Send - Receive MESSAGE with Registration test scenario is completely configured.
Channel#0: MakeRegistration - First Loop Only, Send MESSAGE, Wait 100 Trying or 200 OK for MESSAGE
Channel#1: MakeRegistration - First Loop Only, Wait MESSAGE, Send 200 OK for MESSAGE.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as destination activity.
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VS_017_DUT_SIPv4 IMS MakeCall - ReceiveCall with Registration and RTP
This IMS compliant test runs against a P-CSCF whose IP address needs configured in the SIP Settings page.
The test comprises two activities, Make_Call and Receive_Call, emulating SIP endpoints that establish an IMS-compliant call (Figure 4-12).
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The SIP server you are running this test against needs specified in the Use external server area. Outbound proxy and registrar functionalities on the specified proxy need configured.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity.
Note: The SIP sample templates include another VS_042_DUT_SIPv6 Send - Receive MESSAGE with Registration predefined test identical to this one, except that it uses IPv6 network settings.
Table 4-12. Make_Call Activity Test Settings
Category Settings
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Figure 4-12. VS_017_DUT_SIPv4 IMS MakeCall - ReceiveCall with Registration and RTP
Make_Call establishes an IMS-compliant call, performs a voice session, and ini-tiates call termination. Receive_Call executes the corresponding functions flow on the receiving side.
The Make_Call settings are described in Table 4-13.
Table 4-13. Make_Call Activity Test Settings
Category Settings
Scenario The VS_017_DUT_SIPv4 IMS MakeCall - ReceiveCall with Registration and RTP test scenario comprises 2 channels:
Channel#0: IMS MakeRegistration, IMS Make Call, Voice Session, IMS EndCall Initiate.
Channel#1: IMS MakeRegistration, IMS Receive Call, Voice Session, IMS EndCall Terminate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan Receive_Call activity is configured as call destination.
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VS_018_DUT_SIPv4 IMS Registration with Subscription
This IMS-compliant test runs against a P-CSCF server whose IP address needs configured in the SIP Settings page.
The test comprises only one VoIPSIPPeer activity, Make_Call, that is linked to a scenario channel executing a registration with a SIP server and a subscription operation, as shown in Figure 4-13.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port: Use same value setting from the Execution Settings page.
The P-CSCF you are running this test against must be configured in the Use external server area.
The authentication settings needs configured in the UAC Authentication area.
Codec Settings The default codec settings are used.
RTP Settings Since this test performs media streaming, the Enable media on this activity option is selected.
For this activity, which uses consecutive values for the media IP addresses, RTP ports for all channels can be configured to the 10000 value.
Other Settings The IP version preference is set to IPv4 and the following scenario variables need to be initialized:
• VoIP_Var1: This variable, defined as [310000-], ini-tializes the SIP_Private_Id variable.
• VoIP_Var2: This variable, defined as [310000-], ini-tializes the SIP_Private_Id_Pwd variable.
Note: Receive_Call activity is configured similar to Make_Call, except that it does not specify a call destination since it only terminates a call. Similar to the Make_Call-emulated endpoints, the VoIP_Var1 and VoIPVar2 variables of the Other Settings configuration page are also to initialize the SIP_Private_Id and SIP_Private_Id_Pwd variables.
Note: This test also uses scenario variables to assert the private identity of the SIP UAs, as documented in detail in the notes attached to the test scenario.
Table 4-13. Make_Call Activity Test Settings (Continued)
Category Settings
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Figure 4-13. VS_018_DUT_SIPv4 IMS Registration with Subscription
The Make_Call settings are described in Table 4-14.
Table 4-14. Make_Call Activity Test Settings
Category Settings
Scenario The VS_018_DUT_SIPv4 IMS Registration with Subscription test scenario comprisES 1 channel.
Channel#0: IMS MakeRegistration, IMS Subscribe.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan No VoIPSIPPeer activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port: Use same value setting from the Execution Settings page.
The P-CSCF you are running this test against must be configured in the Use external server area.
The authentication settings must be configured in the UAC Authentication area.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test does not use RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4 and the following scenario variables need to be initialized:
• VoIP_Var1: This variable, defined as [310000-], ini-tializes the SIP_Private_Id variable.
• VoIP_Var2: This variable, defined as [310000-], ini-tializes the SIP_Private_Id_Pwd variable.
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VS_019_DUT_SIPv4 MakeCall - ReceiveCall with RTP - SBC Testing
This test runs against a Session Border Controller (SBC) and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that registers with a SIP server, establishes a call, performs a voice session, and disconnects, as shown in Figure 4-14. Receive_Call executes the test flow on the receiving side.
Figure 4-14. VS_019_DUT_SIPv4 MakeCall - ReceiveCall with Voice - SBC Testing
The Make_Call settings are described in Table 4-15.
Note: This test is particular in that it employs a 1-to-N IP mapping, that is, a single IP is used for all signaling and media endpoints emulated by the Make_Call activity, while endpoints emulated by the Receive_Call activity use N IPs.
Table 4-15. Make_Call Activity Test Settings
Category Settings
Scenario The VS_019_DUT_SIPv4 MakeCall - Receive Call with RTP - SBC Testing test scenario comprises 2 channels.
Channel#0: MakeCall - Authentication, Voice Session (33s), EndCall Initiate.
Channel#1: ReceiveCall, Voice Session (33s), EndCall Terminate.
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The Receive_Call settings are described in Table 4-16.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use same value (per port)
• TCP/UDP/TLS port: Use consecutive values (per port)
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use same value
• UDP port: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as call destination.
SIP Settings The SIP port has the default 5060 setting.
The SBC you are running this test against needs configured in the Use external server area.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test uses RTP streaming, the Enable media on this activity option is selected.
For this activity, whose emulated media channels use the same IP address values, RTP ports need to be configured to the 10000, 10002, 10004, ... sequence.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-16. Receive_Call Activity Test Settings
Category Settings
Scenario The VS_019_DUT_SIPv4 MakeCall - Receive Call with RTP - SBC Testing test scenario comprises 2 channels.
Channel#0: MakeCall - Authentication, Voice Session (33s), EndCall Initiate.
Channel#1: ReceiveCall, Voice Session (33s), EndCall Terminate.
Table 4-15. Make_Call Activity Test Settings
Category Settings
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VS_020_DUT_SIPv4 ReceiveCall - MakeCall with RTP - SBC Testing
This test runs against a Session Border Controller (SBC) and comprises two VoIPSIPPeer activities, Receive_Call and Make_Call.
Receive_Call is linked to a test scenario channel that registers with a SIP server, terminates an incoming call, performs a voice session, and disconnects, as shown in Figure 4-15 on page 4-32. Make_Call executes the call originating functions flow.
Execution Settings The channels are configured to use consecutive IPs from those available in the Network settings.
For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use consecutive values (per port)
• UDP port: Use same value
Dial Plan No activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port: Use same value setting from the Execution Settings page.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test uses RTP streaming, the Enable media on this activity option is selected.
For this activity, whose emulated channels use multiple, consecutive IP address values, RTP ports can be configured to a single 10000 value.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-16. Receive_Call Activity Test Settings
Category Settings
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Figure 4-15. VS_020_DUT_SIPv4 ReceiveCall - MakeCall with RTP - SBC Testing
The Receive_Call settings are given in Table 4-17.
Note: This test uses an N-to-N configuration, that is, Receive_Call and Make_Call both emulate SIP and media endpoints using N distinct IPs.
Table 4-17. Receive_Call Activity Test Settings
Category Settings
Scenario Editor The VS_020_DUT_SIPv4 ReceiveCall - MakeCall with RTP - SBC Testing scenario comprises 2 channels:
Channel#0: ReceiveCall, Voice Session (33s), End Call Initiate
Channel#1: MakeCall - Authentication, Voice Session (33s), EndCall Terminate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use consecutive values (per port)
• UDP port: Use same value
Dial Plan Since this activity terminates a call, no activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port: Use the same value setting from the Execution Settings page.
Codec Settings The codec settings can be left unaltered.
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The Make_Call settings are described in Table 4-18.
RTP Settings Since this test uses RTP streaming, the Enable media on this activity option is selected.
For this activity, which uses consecutive values for the IP addresses, RTP ports for all channels can be configured to the 10000 value.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-18. Make_Call Activity Test Settings
Category Settings
Scenario The VS_020_DUT_SIPv4 ReceiveCall - MakeCall with RTP - SBC Testing scenario comprises 2 channels:
Channel#0: ReceiveCall, Voice Session (33s), End Call Initiate
Channel#1: MakeCall - Authentication, Voice Session (33s), EndCall Terminate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call activity is configured as call destination.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port: Use same value setting from the Execution Settings page.
The SBC you are running this test against is configured in the Use external server area.
Codec Settings The codec settings can be left unaltered.
Table 4-17. Receive_Call Activity Test Settings (Continued)
Category Settings
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VS_021_DUT_SIPv4 MakeCall - EndCall
This test comprises a single VoIPSIPPeer activity, Make_Call, which is linked to a test scenario channel that establishes a signaling-only call with a user-specified device and then disconnects, as shown in Figure 4-16.
Figure 4-16. VS_021_DUT_SIPv4 MakeCall - EndCall Test Scenario
The Make_Call configuration settings are described in Table 4-19.
RTP Settings Since this test uses RTP streaming, the Enable media on this activity option is selected.
For this activity, which uses consecutive values for the media IP addresses, RTP ports for all channels can be configured to the 10000 value.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-19. Make_Call Activity Test Settings
Category Settings
Scenario The VS_021_DUT_SIPv4 MakeCall - EndCall test scenario comprises 1 channel:
Channel#0: MakeCall, Sleep (2s), EndCall
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The IP and port of the call destination device needs specified in this page.
SIP Settings The SIP port has the 5060 default setting.
Codec Settings The codec settings can be left unaltered.
Table 4-18. Make_Call Activity Test Settings
Category Settings
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VS_022_DUT_SIPv4 MakeCall - EndCall with RTP - 33s
This test comprises a single VoIPSIPPeer activity, Make_Call, which is linked to a test scenario channel that establishes a SIP call with media streaming with a user-specified device and then disconnects, as shown in Figure 4-17.
Figure 4-17. VS_022_DUT_SIPv4 MakeCall - EndCall with RTP - 33s Test Scenario
The Make_Call configuration settings are described in Table 4-20.
RTP Settings Since this test does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-20. Make_Call Activity Test Settings
Category Settings
Scenario The VS_022_DUT_SIPv4 MakeCall - EndCall with RTP - 33s test scenario comprises 1 channel:
Channel#0: MakeCall, Voice Session, EndCall Initiate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The IP and port of the call destination device needs specified in this page.
SIP Settings The SIP port has the 5060 default setting.
Codec Settings The codec settings can be left unaltered.
Table 4-19. Make_Call Activity Test Settings
Category Settings
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VS_023_DUT_SIPv4 MakeCall - EndCall with Hold Unhold
This test is similar to the previous VS_021_DUT_SIPv4 MakeCall - EndCall test, except that it executes two additional script functions, SIP Hold and SIP Unhold, before call termination.
Make_Call is linked to the test scenario channel shown in Figure 4-18.
Figure 4-18. VS_023_DUT_SIPv4 MakeCall - EndCall with Hold Unhold Test Scenario
VS_024_DUT_SIPv4 MakeCall - EndCall with SRTP - 33s
This test is similar to the previous VS_022_DUT_SIPv4 MakeCall - EndCall with RTP - 33s test, except that the media traffic is encrypted using SRTP.
VS_025_DUT_SIPv4 MakeCall - EndCall through SIP Redirect Server
This test comprising a single VoIPSIPPeer activity illustrates a call establishment procedure through a SIP redirect server.
RTP Settings The Enable media on this activity option is selected.
For this activity, which uses consecutive values for the media IP addresses, RTP ports for all channels are configured to a single 10000 value.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: The SIP sample templates include a VS_042_DUT_SIPv6 MakeCall - EndCall with Voice - 33s predefined test identical to this one, except that it uses IPv6 network settings.
Note: The test configuration is the same as that of the previous VS_021_DUT_SIPv4 MakeCall - EndCall test.
Table 4-20. Make_Call Activity Test Settings
Category Settings
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Make_Call is linked to a test scenario channel that establishes a call using the SIP MakeCall - Redirect Server procedure capable of resolving 3xx messages and using the first Contact address provided by the redirect server (Figure 4-19).
Figure 4-19. VS_025_DUT_SIPv4 MakeCall - EndCall through Redirect Server Test Scenario
The Make_Call configuration settings are described in Table 4-21.
VS_026_DUT_SIPv4 ReceiveCall - EndCall
This test comprises a single receiving-side VoIPSIPPeer activity linked to a test scenario that answers an incoming signaling-only call and then disconnects, as shown in Figure 4-20.
Table 4-21. Make_Call Activity Test Settings
Category Settings
Scenario The VS_025_DUT_SIPv4 MakeCall - EndCall through Redirect Server test scenario comprises 1 channel.
Channel 0: MakeCall - Redirect Server, Sleep, EndCall Initiate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The IP and port of the call destination device needs specified in this page.
SIP Settings The SIP port has the 5060 default setting for all channels.
Codec Settings The codec settings can be left unaltered, since no media streaming is performed.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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Figure 4-20. VS_026_DUT_SIPv4 ReceiveCall - EndCall Test Scenario
The Receive_Call configuration settings are described in Table 4-22.
VS_027_DUT_SIPv4 ReceiveCall - EndCall with RTP - 33s
This test comprises a single receiving-side VoIPSIPPeer activity linked to a test scenario that answers an incoming call, performs media streaming using the Voice Session script function and then disconnects, as shown in Figure 4-21.
Table 4-22. Receive_Call Activity Test Settings
Category Settings
Scenario The VS_026_DUT_SIPv4 Receive Call - EndCall test scenario comprises 1 channel:
Channel#0: ReceiveCall, Sleep (2s), EndCall Terminate.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan No activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port = Use same value setting from the Execution Settings page.
The SIP server you are running this test against needs configured in the Use external server area.
Codec Settings The codec settings can be left unaltered.
RTP Settings Since this test does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4 and no scenario variables need to be initialized.
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Figure 4-21. VS_027_DUT_SIPv4 ReceiveCall - EndCall with RTP - 33s Test Scenario
The Receive_Call configuration settings are described in Table 4-23.
Table 4-23. Receive_Call Activity Test Settings
Category Settings
Scenario The VS_027_DUT_SIPv4 Receive Call - EndCall with RTP - 33s comprises 1 channel:
Channel#0: SIP Receive Call, Voice Session, SIP End Call Receive.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan No activity is configured as destination activity.
SIP Settings The SIP port has the 5060 default setting for all channels.
The SIP server you are running this test against needs configured in the Use external server area.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected.
The RTP port is configured to the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4 and no scenario variables need to be initialized.
Note: The SIP sample templates include another VS_045_DUT_SIPv6 ReceiveCall - EndCall with RTP - 33s predefined test identical to this one, except that it uses IPv6 network settings.
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VS_028_DUT_SIPv4 ReceiveCall - EndCall with SRTP - 33s
This test is similar to the previous VS_027_DUT_SIPv4 ReceiveCall - EndCall with RTP - 33s test, except the media traffic is encrypted using SRTP.
VS_029_DUT_SIPv4 ReceiveCall - EndCall with Hold Unhold
This test is similar to the previous VS_026_DUT_SIPv4 ReceiveCall - EndCall test, except that it uses another two script functions, SIP Hold and SIP Unhold, before call termination.
Receive_Call is linked to the test scenario channel shown in Figure 4-22.
Figure 4-22. VS_029_DUT_SIPv4 ReceiveCall - EndCall with Hold Unhold Test Scenario
VS_030_B2B_SIPv4_TLS_MakeCall - ReceiveCall - EndCall
This test is similar to the previous VS_001_B2B_SIPv4 MakeCall - ReceiveCall - EndCall test, except that the signaling traffic is encrypted using TLS. SIP end-points emulated by the VoIPSIPPeer activities authenticate each other with cer-tificates (.pem format) included with the IxLoad application.
VS_031_B2B_SIPv4_TLS_ MakeCall - ReceiveCall - EndCall with RTP - 33s
This test is similar to the previous VS_002_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with RTP - 33s test, except that the signaling traffic is encrypted using TLS. SIP endpoints emulated by the VoIPSIPPeer activities authenticate each other with certificates (.pem format) included with the IxLoad application.
VS_032_DUT_SIPv4_TLS_MakeCall - ReceiveCall with Registration
This test is similar to the previous VS_006_DUT_SIPv4 MakeCall - ReceiveCall with Registration test, except that the signaling traffic is encrypted using TLS.
Note: The test configuration is the same as that of the previous VS_026_DUT_SIPv4 ReceiveCall - EndCall test.
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For SIP endpoints authentication in the TLS Settings tab you need to specify a proper certificate.
VS_033_B2B_SIPv4_TLS_ MakeCall - ReceiveCall - EndCall with SRTP - 33s
This test is similar to the previous VS_003_B2B_SIPv4 MakeCall - ReceiveCall - EndCall with SRTP - 33s test, except that the signaling traffic is encrypted using TLS. SIP endpoints emulated by the VoIPSIPPeer activities authenticate each other with certificates (.pem format) included with the IxLoad application.
SIPv4_UDP_Basic_Call_without_RTP_Max_CPS
This is a performance test that runs in Back-to-Back mode and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that establishes a basic SIP call without media streaming and then disconnects. Receive_Call executes the test flow on the receiving side.
The configured test objective is 700 CPS.
SIPv4_TCP_Basic_Call_without_RTP_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_without_RTP_Max_CPS test, except that the signaling traffic is sent over TCP.
The configured test objective is 600 CPS.
SIPv4_TLS_Basic_Call_without_RTP_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_without_RTP_Max_CPS test, except that the signaling traffic is sent using TLS.
The configured test objective is 600 CPS.
Note: Further sample test configurations with indexes up to VS_050 that are provided in the IxLoad installation kit represent configurations similar to those listed above, except that they are configured using IPv6 instead of IPv4 settings.
Note: The following tests specially created for use with Acceleron load module boards are stored both in the VoIPSIP\Acceleron_NA and the VoIPSIP\Acceleron_10G folders of the IxLoad Getting Started window.
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SIPv4_UDP_Basic_Call_with_RTP_1sec_Max_CPS
This is a performance test that runs in Back-to-Back mode and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that establishes a basic SIP call with media streaming (using the Voice Session script function) and then discon-nects. Receive_Call executes the test flow on the receiving side.
The configured test objective is 500 CPS.
SIPv4_TCP_Basic_Call_with_RTP_1sec_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_1sec_Max_CPS test, except that the signal-ing traffic is sent over TCP.
The configured test objective is 500 CPS.
SIPv4_TLS_Basic_Call_with_RTP_1sec_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_1sec_Max_CPS test, except that the signal-ing traffic is sent using TLS.
The configured test objective is 250 CPS.
SIPv4_UDP_Basic_Call_with_RTP_30sec_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_1sec_Max_CPS test, except that the media is played for a duration of 30 seconds.
The configured test objective is 200 CPS.
SIPv4_TCP_Basic_Call_with_RTP_30sec_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_30sec_Max_CPS test, except that the sig-naling traffic is sent over TCP.
The configured test objective is 200 CPS.
SIPv4_TLS_Basic_Call_with_RTP_30sec_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_30sec_Max_CPS test, except that the sig-naling traffic is sent using TLS.
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The configured test objective is 200 CPS.
SIPv4_UDP_Basic_Call_with_RTP_3min_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_1sec_Max_CPS test, except that the media is played for a duration of 3 minutes.
The configured test objective is 40 CPS.
SIPv4_TCP_Basic_Call_with_RTP_3min_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_3min_Max_CPS test, except that the signal-ing traffic is sent using TLS.
The configured test objective is 40 CPS.
SIPv4_TLS_Basic_Call_with_RTP_3min_Max_CPS
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_3min_Max_CPS test, except that the signal-ing traffic is sent over TCP.
The configured test objective is 40 CPS.
SIPv4_UDP_Basic_Call_with_RTP_Max_Sessions
This is a performance test that runs in Back-to-Back mode and comprises two VoIPSIPPeer activities, Make_Call and Receive_Call.
Make_Call is linked to a test scenario channel that establishes a basic SIP call with media streaming (using the Voice Session script function) and then discon-nects. Receive_Call executes the test flow on the receiving side.
The configured test objective is 8000 Channels.
SIPv4_TCP_Basic_Call_with_RTP_Max_Sessions
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_Max_Sessions test, except that the media is sent using SRTP.
The configured test objective is 8000 Channels.
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SIPv4_TLS_Basic_Call_with_RTP_Max_Sessions
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_Max_Sessions test, except that the media is sent using SRTP.
The configured test objective is 8000 Channels.
SIPv4_UDP_Basic_Call_with_SRTP_Max_Sessions
This test is similar to the previous SIPv4_UDP_Basic_Call_with_RTP_Max_Sessions test, except that the media is sent using SRTP.
The configured test objective is 300 Channels.
SIPv4_TCP_Basic_Call_with_SRTP_Max_Sessions
This test is similar to the previous SIPv4_TCP_Basic_Call_with_RTP_Max_Sessions test, except that the media is sent using SRTP.
The configured test objective is 300 Channels.
SIPv4_TLS_Basic_Call_with_SRTP_Max_Sessions
This test is similar to the previous SIPv4_TLS_Basic_Call_with_RTP_Max_Sessions test, except that the media is sent using SRTP.
The configured test objective is 200 Channels.
SIPv4_Proxy
This test illustrates the case of SIP UAs that register with a Registrar and then establish calls via an SIP Proxy server, whereby all test entities – SIP UAs, Reg-istrar and Proxy server – are emulated by IxLoad activities. During the call estab-lishment phase, the Proxy stays is the message path, while after the call established media is exchanges directly between SIP endpoints.
Caller and Callee endpoints are emulated by the Make_Call and Receive_Call VoIPSIPPeer activities; the Proxy / Registrar is simulated by four VoIPSIPPeer activities having an associated VoIPSIPCloud activity (Figure 4-23).
Note: The following two tests are stored in the VoIPSIP\Proxy_Scenarios folder of the IxLoad Getting Started window.
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Figure 4-23. SIP UAs, Proxy/Registrar Emulation
The executed protocol flow is the following:
• Caller executes the SIP protocol flow of the calling endpoints, including the endpoint registration/de-registration operations at the start and the end of the test flow. On the established call, a Talk script function plays media directly between Caller and Callee. Finally Caller terminates the call using an End Call Initiate procedure.
• Callee executes the SIP protocol flow of the calling endpoints, including the endpoint registration/de-registration operations at the start and the end of the test flow. On the established call, a Talk script function plays media directly between Callee and Caller.
• R_Caller, R_Callee handle the registration / de-registration for the Caller and Callee endpoints respectively.
• Wait_Call handles the incoming call from the Proxy server perspective by communicating with the Caller: an initial INVITE Processing procedure waits for an incoming INVITE message, then sends the response using a Send 180 and 200 Ok procedure.
Note that inside the INVITE Processing procedure, the Wait INVITE pro-cedure extracts the value of some message headers, which are then used by the subsequent Variable Set function to initialize a number of scenario vari-ables. These variables are eventually used by the Send INVITE procedure of the Make_Call activity for initializing the values of INVITE message head-ers.
Using a Wait procedure, the scenario execution is then paused for a duration of 1 second, providing a time window for the SIP UAs to exchange media directly, without the Proxy staying in the media path.
Finally, a SIP EndCall - Receiving Channel procedure waits for an incom-ing BYE message from Caller, sets the E(nd)C(all)R(eceived) scenario vari-able to ‘1’ (‘true’), and responds with a 200 Ok message. In case no BYE message was received, the SIP EndCall - Receiving Channel procedure exits on the Timeout output and checks whether a BYE message was received by the Make_Call activity (test scenario channel 2) by testing the ECR value for that channel.
• Make_Call handles the outgoing call from the Proxy server perspective by communicating with the Callee: When an INVITE message is received by Wait_Call, this is retransmitted using a Send INVITE procedure, while sub-sequent procedures wait for responses to the INVITE and send an ACK.
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Note that the Send INVITE procedure uses a number of scenario variables initialized within the INVITE Processing procedure for passing appropriate values to the message headers of the sent INVITE message.
Using a Wait procedure, the scenario execution is then paused for a duration of 1 second, providing a time window for the SIP UAs to exchange media directly, without the Proxy staying in the media path.
Finally, a SIP EndCall - Calling Channel procedure waits for a BYE mes-sage, sets the E(nd)C(all)R(eceived) scenario variable to ‘1’ (‘true’), and completes the call. In case no BYE message was received, the SIP EndCall - Calling Channel procedure exits on the Timeout output and checks whether a BYE message was received by the Wait_Call activity (test scenario channel 1) by testing the ECR value for that channel.
SIPv4_B2BUA
This test is similar to the previous SIPv4_Proxy test, except that it emulates a B2B User Agent (UA) instead of a Proxy server.
VoIPSIP with MSRP The tests from this category emulate SIP UAs that establish SIP sessions, then exchange text or transfer files using the MSRP protocol; some tests are also pro-vided that transmit voice over SIP calls and simultaneously negotiate other SIP sessions for the sending of text or files using MSRP.
All test from this category run in Back-to-Back mode and contain two VoIPSIP-Peer activities, Make_Call and Receive_Call.
MSRP_01_SIPv4_UDP_Text_Bidirectional
Make_Call is linked to a test scenario channel that establishes a SIP call, sends a text message using the MSRP Session script function, and then disconnects, as shown in Figure 4-1. Receive_Call executes the test flow on the receiving side, whereby it uses a corresponding MSRP Session script function to receive and also send a text message over the established session (bidirectional text transfer).
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Figure 4-24. MSRP_01_SIPv4_UDP_Text_Bidirectional Test Scenario
The Make_Call configuration settings are described in Table 4-24.
Table 4-24. Make_Call Activity Test Settings
Category Settings
Scenario The test scenario comprises 2 channels executing a basic call procedure with media exchange.
Channel#0: MakeCall procedure, MSRP Session function, End Call Initiate procedure.
Channel#1: ReceiveCall procedure, MSRP Session function, End Call Receive procedure.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port).
Dial Plan The source phone numbers are specified using the 160[00000000-] regular expression.
The Receive_Call activity and port 5060 are configured as call destination.
SIP Settings The 5060 default SIP listening port setting is configured for all channels.
Codec Settings The default codec settings are used.
RTP Settings Since no RTP media is streamed by the emulated SIP endpoints, the Enable media on this activity option is not selected.
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MSRP_02_SIPv4_UDP_File_Transfer_Uni_aSDP
Make_Call is linked to a test scenario channel that establishes a SIP call and negotiates file transfer parameters using a SIP Make Call procedure, sends a synthetic file using the MSRP Session script function, and then disconnects using a SIP End Call Initiate procedure, as shown in Figure 4-1. The INVITE function contained in the SIP Make Call procedure uses an automatic SDP (aSDP) offer.
Receive_Call executes a similar test flow on the receiving side, whereby its MSRP Session script function only receives the incoming data (unidirectional file transfer).
The Make_Call configuration settings are described in Table 4-24.
MSRP settings The MSRP endpoints are specified using the 160[00000000-].example.com sequence generator expression.
The text message to be transmitted is configured in the Content page of the MSRP Session function.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call. The MSRP Session function that is executed by the Receive-Call activity also specifies a text message to be transmitted.
Table 4-25. Make_Call Activity Test Settings
Category Settings
Scenario The test scenario comprises 2 channels executing a basic call procedure with media exchange.
Channel#0: MakeCall procedure, MSRP Session function, EndCall Initiate procedure.
Channel#1: ReceiveCall procedure, MSRP Session function, EndCall Receive procedure.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port).
Table 4-24. Make_Call Activity Test Settings (Continued)
Category Settings
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MSRP_03_SIPv4_UDP_File_Transfer_Uni_customSDP
Make_Call is linked to a test scenario channel that establishes a call using a SIP Make Call procedure, sends a synthetic file using the MSRP Session script function, and then disconnects using a SIP End Call Initiate procedure, as shown in Figure 4-1. The INVITE function contained in the SIP Make Call pro-cedure uses a custom SDP (cSDP) offer, whereby the SDP file-selector attribute is defined using the predefined VoIP_MSRPFile0, VoIP_MSRPFileType0, VoIP_MSRPFileSize0, and VOIP_MSRPFileHash0 variables, which specify the transmitted file configured on the MSRP Session function. Another SDP attri-bute, file-transfer-id, is initialized using the Ixload predefined generateguid func-tion.
Receive_Call executes a similar test flow on the receiving side, whereby its MSRP Session script function only receives the incoming file (unidirectional transfer).
The activity-level settings for this test are the same as those for the previous MSRP_02_SIPv4_UDP_File_Transfer_Uni_aSDP test.
Dial Plan The source phone numbers are specified using the 160[00000000-] regular expression.
The Receive_Call activity and port 5060 are configured as call destination.
SIP Settings The 5060 default SIP listening port setting is configured for all channels.
Codec Settings The default codec settings are used.
RTP Settings Since no RTP media is streamed by the emulated SIP endpoints, the Enable media on this activity option is not selected.
MSRP settings The MSRP endpoints are specified using the 160[00000000-].example.com sequence generator expression.
The transmitted synthetic file is configured in the Content page of the MSRP Session function.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
Table 4-25. Make_Call Activity Test Settings (Continued)
Category Settings
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MSRP_04_SIPv4_UDP_Simultaneous_File_Text_ Transfer
This test performs the simultaneous sending of a text message and of a synthetic file from the Make_Call activity to the Receive_Call activity.
For each media to be transmitted, text and file, the Make_Call activity executes SIP Make Call procedures that establish the call and negotiate media parame-ters, then different MSRP Session functions are called for transmitting the file and the text.
Following the first SIP Make Call which initiates the session for file transfer, a MSRP Session function is executed that sends a synthetic file. This MSRP Session function is configured in a non-blocking mode (background execution mode), so execution advances to a second SIP Make Call procedure, which initi-ates the session for the sending of text using a second MSRP Session function. This results in the two MSRP Session functions transmitting simultaneously. A first SIP End Call procedure eventually terminates the session for the sending of text.
Since the first MSRP Session function is configured in non-blocking mode, a MSRP Control script function is used to probe if the file transfer completed or not; eventually, when the MSRP file transmission is found to have completed, a second SIP End Call is called for terminating the file transfer session. This func-tions flow is illustrated in Figure 4-25:
Figure 4-25. MSRP_04_SIPv4_UDP_Simultaneous_File_Text_Transfer Test Scenario
Receive_Call executes a similar test flow on the receiving side, whereby its MSRP Session script functions receives the incoming text and file data.
MSRP_05_SIPv4_UDP_Simultaneous_Voice_Text_Transfer
This test performs the simultaneous sending of audio (voice) and of a text mes-sage from the Make_Call activity to the Receive_Call activity.
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For each media to be transmitted, voice and text, the Make_Call activity executes two SIP Make Call procedures, whereby each contained INVITE function sends a media-specific custom SDP offer using VoIP predefined variables. Once the SIP calls have been established, Voice Session and MSRP Session functions are called for transmitting the voice and the text simultaneously.
Following the first SIP Make Call that initiates the SIP session for speech, a Voice Session function is executed that plays an audio file. This Voice Session function is configured in a non-blocking mode (background execution mode) at activity level, so execution advances to a second SIP Make Call procedure, which initiates the session for the sending of text using a MSRP Session func-tion. This results in the Voice Session and the MSRP Session functions transmit-ting simultaneously. A first SIP End Call procedure eventually terminates the session for the sending of text using MSRP.
Since the Voice Session function is configured in non-blocking mode at activity level, a RTP Control script function is used to probe if the transmission of speech completed or not; eventually, when the sending of speech is found to have completed, a SIP End Call Initiate is executed for terminating the voice session.
The Receive_Call activity executes a similar test flow on the receiving side, whereby its MSRP Session script function both receives and sends a text mes-sage over the same established session.
This functions flow is illustrated in Figure 4-26:
Figure 4-26. MSRP_05_SIPv4_UDP_Simultaneous_Voice_Text_Transfer Test Scenario
Receive_Call executes a similar test flow on the receiving side, whereby its Voice Session function receives the voice data sent by Make_Call, while the MSRP Session function receives the incoming text data.
The Make_Call configuration settings are described in Table 4-26.
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Table 4-26. Make_Call Activity Test Settings
Category Settings
Scenario The test scenario comprises 2 channels executing the afore-mention functions flow:
Channel#0: SIP Make Call procedure, Voice Session function, SIP Make Call procedure, MSRP Session function, EndCall Initiate procedure, RTP Control function, EndCall Initiate function.
Channel#1: SIP Receive Call procedure, Voice Session function, SIP Receive Call procedure, MSRP Session function, EndCall Receive procedure, RTP Control function, EndCall Receive function.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port).
Dial Plan The source phone numbers are specified using the 160[00000000-] regular expression.
The Receive_Call activity and port 5060 are configured as call destination.
SIP Settings The 5060 default SIP listening port setting is configured for all channels.
Codec Settings The default G711 aLaw and uLaw codecs are used.
Audio An audio clip is configured to be played for a duration of 30 seconds.
RTP Settings The Enable media on this activity option is selected for media streaming to be performed.
The non-blocking behavior is selected for RTP Session script function.
MSRP settings The MSRP endpoints are specified using the 160[00000000-].example.com sequence generator expression.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity, since it only terminates a call.
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MSRP_06_SIPv4_UDP_Simultaneous_Voice_File_Transfer
This test is similar to the MSRP_05_SIPv4_UDP_Simultaneous_Voice_Text_ Transfer one, with the difference that the MSRP Session function from chan-nel#0 transmits a synthetic file instead of a text message. The corresponding MSRP Session function from channel#1 receives the file and does not transmit anything.
VoIPSIP Cloud Test Configurations
This section provides a brief description of the sample VoIPSIP Cloud test con-figuration files contained in the IxLoad installation kit.
For these tests, VoIPSIP Peer activities are assigned to VoIPSIP Clouds that emulate a number of SIP Proxy servers.
VoIPSIP Cloud test configurations are based on the topologies shown in Figure 4-27 and Figure 4-28.
Figure 4-27. SIP Cloud UAs Opposite SIP UAs
Note: Sample SIP test configurations do not have retransmissions enabled at activity level. For complex tests that require retransmissions you can enable retransmissions at activity level.
In this topology, SIP UAs on one side, emulated by a VoIPSIP Peer activity, are configured as part of a SIP cloud, whereas remote UAs are not part of a cloud.
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Figure 4-28. SIP CLoud UAs Opposite SIP Cloud UAs
SC_001_B2B_SIPv4_T1_MakeCall_from_Cloud
This test, based on the topology shown in Figure 4-27, comprises a VoIPSIP Peer and a VoIPSIPCloud activity on the call originating side and a VoIPSIP Peer activity on the terminating side. The DUT shown in Figure 4-27 is actually not present in the test configuration, the test being run in Back-to-Back mode.
The SIP endpoints emulated by Make_Call and configured into the SIP cloud establish signaling-only calls with the endpoints emulated by Receive_Call (Figure 4-29).
Figure 4-29. SC_001_B2B_SIPv4_T1_BasicCall
The Make_Call configuration settings are described in Table 4-27.
In this topology, both the SIP UAs on the call originating side and those on the call terminating side, emulated by VoIPSIP Peer activities, are configured into a SIP cloud.
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SC_002_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_n_to_n
This test replicates the topology shown in Figure 4-27 and comprises a VoIPSIP Peer and a VoIPSIPCloud activity on the call originating side, and a VoIPSIP-Peer activity on the terminating side. The DUT shown in Figure 4-27 is actually not present in the test configuration, the test being run in Back-to-Back mode.
The SIP UAs emulated by Make_Call and configured in the SIP cloud establish signaling and media calls with the endpoints emulated by Receive_Call (Figure 4-30).
Table 4-27. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_001_B2B_SIPv4_T1_BasicCall test scenario comprises a basic call originating sequence on the first channel, and a call terminating sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as call destination.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings Make_Call is assigned to VoIPSIPCloud1 which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity. Also, Receive_Call does not use SIP cloud emulation.
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Figure 4-30. SC_002_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_n_to_n
The Make_Call configuration settings are described in Table 4-28.
Table 4-28. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_002_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_n_to_n test scenario comprises a basic call originating functions sequence on the first channel, and a call terminating sequence on the second channel. After call establishment, media is exchanged directly between the endpoints using the VoiceSession function.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call activity is configured as call destination.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings Make_Call is assigned to VoIPSIPCloud1 which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
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SC_003_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_1_to_n
This test is similar to the previous SC_002_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_n_to_n test, except that media endpoints on the originating activity are configured using a single IP address (which differs from the signaling IP addresses) and consecutive UDP ports.
SC_004_B2B_SIPv4_T1_MakeCall_from_Cloud_w_Registration
This test replicating the topology shown in Figure 4-27 comprises two VoIPSIP Peer (Make_Call and Registrar) and a VoIPSIP Cloud activity on the call origi-nating side, and a VoIPSIPPeer activity (Receive_Call) on the terminating side. The DUT shown in Figure 4-27 is actually not present in the test configuration, the test being run in Back-to-Back mode.
After registering with the Registrar server emulated by the Registrar activity, the SIP UAs emulated by the Receive_Call activity terminate incoming signaling-only calls originated by the endpoints emulated by the Make_Call activity (Figure 4-32).
Figure 4-31. SC_004_B2B_SIPv4_T1_MakeCall_from_Cloud_w_Registration
RTP Settings The Enable media on this activity option is selected.
The RTP port uses the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity. Also, Receive_Call does not use SIP cloud emulation.
Table 4-28. Make_Call Activity Test Settings
Category Settings
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Both the Make_Call and the Registrar VoIPSIP Peer activities are assigned to VoIPSIPCloud1.
The Make_Call configuration settings are described in Table 4-33.
Table 4-29. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_004_B2B_SIPv4_T1_MakeCall_from_Cloud_w_Registration test scenario comprises a basic call originating sequence on the second channel, and a call terminating sequence on the third channel. Before receiving the call, terminating endpoints first execute a registration with the Registrar-emulated server.
The first channel function flow, linked to the Registrar activity, responds to incoming registration requests sent by endpoints emulated by the Receive_Call activity.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1.
Two emulated servers defined in the cloud, sip_server1 and sip_server2, are configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and the VoIP_IPAddress0 variable is initialized to a range of IP addresses that are assigned to endpoints emulated by Receive_Call activity. The variable is used on the Send INVITE script function to specify the call destination.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity. Also, Receive_Call does not use SIP cloud emulation.
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The Receive_Call configuration settings are described in Table 4-33.
The Registrar configuration settings are described in Table 4-31.
Table 4-30. Receive_Call Activity Test Settings
Category Settings
Scenario Editor The SC_004_B2B_SIPv4_T1_MakeCall_from_Cloud_w_Registration test scenario comprises a basic call originating sequence on the second channel, and a call terminating sequence on the third channel. Before receiving the call, terminating endpoints first execute a registration with the Registrar-emulated server.
The first channel function flow, linked to the Registrar activity, responds to incoming registration requests sent by endpoints emulated by the Receive_Call activity.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan No activity is configured as call destination.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The registrar server is specified in the Use external server area.
Cloud SIP Settings This activity is not assigned to any SIP cloud.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-31. Registrar Activity Test Settings
Category Settings
Scenario Editor The SC_004_B2B_SIPv4_T1_MakeCall_from_Cloud_w_Registration test scenario comprises a basic call originating sequence on the second channel, and a call terminating sequence on the third channel. Before receiving the call, terminating endpoints first execute a registration with the Registrar-emulated server.
The first channel function flow, linked to the Registrar activity, responds to incoming registration requests sent by endpoints emulated by the Receive_Call activity.
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The VoIPSIPCloud1 configuration settings are described in Table 4-32.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan Phone numbers are specified using the 170[00000000-] formula.
Since this activity only receives incoming registration requests, no call destination needs configured.
SIP Settings The SIP port has the default setting [5070-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1.
The activity has an overriding dispatching rule defined, specifying that incoming SIP Register request messages be dispatched based on the 170[00000000-] formula.
The Use server option is enabled for the sip_server#3 associated with the emulated Registrar server.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-32. VoIPSIPCloud1 Activity Test Settings
Category Settings
Settings The SIP cloud emulates three SIP Proxy servers, sip_server#1, sip_server#2, and sip_server#3.
Preview Cloud Traffic
The Make_Call and Registrar activities in the cloud are shown to execute the first and second channels of the SC_004_B2B_SIPv4_T1_MakeCall_from_Cloud_w_Registration test scenario.
Table 4-31. Registrar Activity Test Settings
Category Settings
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SC_005_B2B_SIPv4_T1_ReceiveCall_by_Cloud
This test replicating the topology shown in Figure 4-27 comprises a VoIPSIP Peer and a VoIPSIPCloud activity on the call terminating side, and a VoIPSIP-Peer activity on the originating side. The DUT shown in Figure 4-27 is actually not present in the test configuration, the test being run in Back-to-Back mode.
The SIP UAs emulated by Make_Call establish signaling-only calls with the end-points emulated by Receive_Call and configured into the SIP cloud (Figure 4-32).
Figure 4-32. SC_005_B2B_SIPv4_T1_ReceiveCall_by_Cloud
The Make_Call configuration settings are described in Table 4-33.
Table 4-33. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_005_B2B_SIPv4_T1_ReceiveCall_by_Cloudtest scenario comprises a basic call originating functions sequence on the second channel, and a call terminating sequence on the first channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The SIP server you are running this test against needs specified in the Use external server area, with an outbound proxy functionality selected.
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SC_006_B2B_SIPv4_T1_ReceiveCall_by_Cloud_RTP_n_to_n
This test replicating the topology shown in Figure 4-27 comprises a VoIPSIP Peer and a VoIPSIPCloud activity on the call terminating side, and a VoIPSIP-Peer activity on the originating side. The DUT shown in Figure 4-27 is actually not present in the test configuration, the test being run in Back-to-Back mode.
The SIP UAs emulated by Make_Call establish media calls with the endpoints emulated by Receive_Call and configured into the SIP cloud (Figure 4-33).
Figure 4-33. SC_006_B2B_SIPv4_T1_ReceiveCall_by_Cloud_RTP_n_to_n
The Make_Call configuration settings are described in Table 4-34.
Cloud SIP Settings This activity is not configured as part of any SIP cloud.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity.
Receive_Call is assigned to VoIPSIPCloud1 which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Table 4-33. Make_Call Activity Test Settings
Category Settings
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Table 4-34. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_006_B2B_SIPv4_T1_ReceiveCall_by_Cloud_RTP_n_to_n test scenario comprises a basic call originating functions sequence on the first channel, and a call terminating sequence on the second channel.
After call establishment, media is exchanged using the VoiceSession function.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The SIP server you are running this test against needs specified in the Use external server area, with outbound proxy functionality configured.
Cloud SIP Settings This activity is not configured as part of any SIP cloud.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected.
The RTP port uses the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity.
Receive_Call is configured as part of VoIPSIPCloud1 which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
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SC_007_B2B_SIPv4_T1_Receive_Call_by_Cloud_RTP_n_to_1
This test is similar to the previous SC_006_B2B_SIPv4_T1_Receive_Call_by_Cloud_RTP_n_to_n test, except for the RTP channels configuration on the Receive_Call activity, which uses the same IP address (different from the signal-ling IP address) and consecutive UDP ports, resulting in a media configuration as shown in Figure 4-34 below.
Figure 4-34. N-to-1 Media Configuration
Note: Since both originating and terminating side media channels are configured using multiple IP addresses, the test uses a n-to-n media configuration as shown in the figure below:
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SC_008_DUT_SIPv4_T1_MakeCall_from_Cloud_w_Reg_to_DUT
This test replicating the topology shown in Figure 4-27 comprises a VoIPSIP Peer and a VoIPSIPCloud activity on the call terminating side, and a VoIPSIP-Peer activity on the originating side.
After registering with a SIP Registrar server (DUT), the SIP UAs emulated by Receive_Call terminate signaling-only calls originated by the Make_Call-emu-lated endpoints (Figure 4-35).
Figure 4-35. SC_008_DUT_SIPv4_T1_MakeCall_from_Cloud_w_Reg_to_DUT
The Make_Call configuration settings are described in Table 4-35.
Table 4-35. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_008_DUT_SIPv4_T1_MakeCall_from_Cloud_w_Reg_to_DUT test scenario comprises a basic call originating functions sequence on the first channel, and a call terminating sequence on the second channel. Prior to accepting incoming calls, endpoints executing the second channel functions flow first register with a Registrar server.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as destination activity.
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SC_009_DUT_SIPv4_T1_MakeCall_from_Cloud_w_Reg_to_DUT_RTP
This test is similar to the previous SC_008_B2B_SIPv4_T1_MakeCall_from_Cloud_with_Reg_to_DUT test, except that media is also exchanged on the established calls.
Media endpoints on the originating activity are configured using a single IP address and consecutive UDP ports. Media endpoints on the terminating end-points are configured using consecutive IPs and the same port.
SC_010_B2B_SIPv4_T1_MakeCall_ReceiveCall_IM_from_Cloud_RTP
This test replicating the topology shown in Figure 4-27 comprises on one side a VoIPSIPCloud that has three associated VoIPSIPPeers, and three VoIPSIPPeers on the other side. The DUT shown in Figure 4-27 is actually not present in the test configuration, the test being run in Back-to-Back mode.
Make_Call_UDP- and Receive_Call_TCP-emulated endpoints establish and ter-minate basic signaling and media calls with Receive_Call_UDP- and Make_Call_TCP-emulated endpoints, respectively.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The SIP server (DUT) you are running this test against needs specified in the Use external server area, with outbound proxy functionality selected.
Cloud SIP Settings This activity is configured into the VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling-only calls, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity. The SIP server specified in the Use external server area has outbound proxy and registrar functionality selected.
Table 4-35. Make_Call Activity Test Settings
Category Settings
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IM_Send_Receive- and IM_Receive_Send- emulated endpoints exchange instant messages both ways.
Each pair of traffic source and destination are configured into an ActivityLink having a corresponding 2 channel test scenario.
The Make_Call_UDP configuration settings are described in Table 4-36.
Table 4-36. Make_Call_UDP Activity Test Settings
Category Settings
Scenario Editor The SC_010_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP tst test scenario comprises a basic call originating sequence on the first channel, and a call terminating sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call_UDP activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling and media calls, the Enable media on this activity option is selected.
The RTP port is configured to the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call_UDP uses the same settings as Make_Call_UDP, except for the Dial Plan page, which does not need to specify a destination activity. Also, Receive_Call_UDP is not assigned to any cloud.
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The Make_Call_TCP configuration settings are described in Table 4-37.
Table 4-37. Make_Call_TCP Activity Test Settings
Category Settings
Scenario Editor The SC_010_B2B_SIPv4_T1_ReceiveCall_by_Cloud_RTP test scenario comprises a basic call originating sequence on the first channel, and a call terminating sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call_TCP activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The Override transport option is enabled and TCP transport is selected.
Cloud SIP Settings This activity is not assigned to any cloud.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling and media calls, the Enable media on this activity option is selected.
The RTP port is configured to the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call_TCP uses the same settings as Make_Call_TCP, except for the Dial Plan page, which does not need to specify a destination activity. Also, Receive_Call_UDP is assigned to VoIPSIPCloud1 which emulates two Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
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The IM_Send_Receive configuration settings are described in Table 4-38.
SC_011_B2B_SIPv4_T2_BasicCall_between_two_Clouds
This test replicating the topology shown in Figure 4-28 comprises a VoIPSIP Peer assigned to a VoIPSIPCloud both on the call originating and the call termi-nating side. The DUT shown in Figure 4-28 is actually not present in the test con-figuration, the test being run in Back-to-Back mode.
The SIP UAs emulated by Receive_Call terminate signaling-only calls originated by the Make_Call-emulated endpoints (Figure 4-36).
Table 4-38. IM_Send_Receive Activity Test Settings
Category Settings
Scenario Editor The SC_010_B2B_SIPv4_T1_Instant_Messaging test scenario comprises a basic instant messaging send and receive sequence on the first channel, and a receive and send sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The IM_Receive_Send activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints perform only instant messaging procedures, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: IM_Receive_Send uses the same settings as IM_Send_Receive, except for the Dial Plan page, which specifies IM_Send_Receive as destination activity. Also, IM_Receive_Send is not assigned to any cloud.
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Figure 4-36. SC_011_B2B_SIPv4_T2_BasicCall_between_two_Clouds
The Make_Call configuration settings are described in Table 4-39.
Table 4-39. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_011_B2B_SIPv4_T2_BasicCall_between_two_Clouds test scenario comprises a basic call originating functions sequence on the first channel, and a call terminating sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The Receive_Call activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling-only calls, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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SC_012_B2B_SIPv4_T2_BasicCall_between_two_Clouds_no_Routes
This test is similar to the previous C_011_B2B_SIPv4_T2_BasicCall_between_two_Clouds one, except that no SIP message routing functionality is configured at activity level (the Keep in route option in Cloud SIP Settings page is not selected).
SC_013_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n
This test replicating the topology shown in Figure 4-28 comprises a VoIPSIP Peer assigned to a VoIPSIPCloud both on the call originating and the call termi-nating side. The DUT shown in Figure 4-28 is actually not present in the test con-figuration, the test being run in Back-to-Back mode.
The SIP UAs emulated by Receive_Call terminate signaling and media calls orig-inated by the Make_Call-emulated endpoints (Figure 4-37). After calls are estab-lished, media traffic is exchanged directly between endpoints
Figure 4-37. SC_013_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n
The Make_Call configuration settings are described in Table 4-39.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity.
Table 4-40. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_013_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n test scenario comprises a basic call originating functions sequence on the first channel, and a call terminating sequence on the second channel.
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Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling and media calls, the Enable media on this activity option is selected.
The RTP port is configured to the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity.
Table 4-40. Make_Call Activity Test Settings
Category Settings
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SC_014_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_1_to_1
This test is similar to the previous SC_013_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n one, except that media is exchanged between endpoints by sharing a single IP address (different than the signaling ones) and consecutive values for the UDP ports.
SC_015_DUT_SIPv4_T2_BasicCall_between_two_Clouds
This test replicating the topology shown in Figure 4-28 comprises a VoIPSIP Peer assigned to a VoIPSIPCloud both on the call originating and the call termi-nating side.
The SIP UAs emulated by Receive_Call terminate signaling-only calls originated by the Make_Call-emulated endpoints (Figure 4-38).
Figure 4-38. SC_015_DUT_SIPv4_T2_BasicCall_between_two_Clouds
The Make_Call configuration settings are described in Table 4-41.
Table 4-41. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_015_DUT_SIPv4_T2_BasicCall_between_two_Clouds test scenario comprises a basic call originating sequence on the first channel, and a call terminating sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
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SC_016_DUT_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n
This test replicating the topology shown in Figure 4-28 comprises a VoIPSIP Peer assigned to a VoIPSIPCloud both on the call originating and the call termi-nating side.
The SIP UAs emulated by Receive_Call terminate signaling and media calls orig-inated by the Make_Call-emulated endpoints (Figure 4-39).
Dial Plan The Receive_Call activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The SIP server (DUT) you are running this test against needs specified in the Use external server area, with outbound proxy functionality selected.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling and media calls, the Enable media on this activity option is selected.
The RTP port is configured to the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity.
Table 4-41. Make_Call Activity Test Settings
Category Settings
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Figure 4-39. SC_016_DUT_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n
The Make_Call configuration settings are described in Table 4-42.
Table 4-42. Make_Call Activity Test Settings
Category Settings
Scenario Editor The SC_016_DUT_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n test scenario comprises a basic call originating sequence on the first channel, and a call terminating sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The SIP server (DUT) you are running this test against needs specified in the Use external server area, with outbound proxy functionality selected.
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SC_017_DUT_SIPv4_T2_BasicCall_between_two_Clouds_RTP_1_to_1
This test is similar to the previous SC_017_DUT_SIPv4_T2_BasicCall_between_two_Clouds_RTP_n_to_n one, except that media is exchanged between endpoints sharing a single IP address (different than the signaling ones) and con-secutive values for the UDP ports.
SC_018_B2B_SIPv4_T2_MakeCall_ReceiveCall_IM_w_RTP_two_Clouds
This test replicating the topology shown in Figure 4-28 comprises on each side a VoIPSIPCloud having three associated VoIPSIP Peers. The DUT shown in Figure 4-28 is actually not present in the test configuration, the test being run in Back-to-Back mode.
Make_Call_UDP and Receive_Call_TCP emulated endpoints on VoIPSIPCloud1 establish and terminate basic signaling and media calls with Receive_Call_UDP and Make_Call_TCP endpoints on VoIPSIPCloud2, respec-tively.
IM_Send_Receive emulated endpoints on VoIPSIPCloud1 and IM_Receive_Send on VoIPSIPCloud2 exchange instant messages both ways.
Each pair of traffic source and destination are configured into an ActivityLink having a corresponding test scenario.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling and media calls, the Enable media on this activity option is selected.
The RTP port is configured to the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call uses the same settings as Make_Call, except for the Dial Plan page, which does not need to specify a destination activity.
Table 4-42. Make_Call Activity Test Settings
Category Settings
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The Make_Call_UDP configuration settings are described in Table 4-43.
Table 4-43. Make_Call_UDP Activity Test Settings
Category Settings
Scenario Editor The SC_018_B2B_SIPv4_T2_MakeCall_w_RTP_two_Clouds tst test scenario comprises a basic call originating sequence on the first channel, and a call terminating sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call_UDP activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling and media calls, the Enable media on this activity option is selected.
The RTP port is configured to the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call_UDP uses the same settings as Make_Call_UDP, except for the Dial Plan page, which does not need to specify a destination activity.
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The Make_Call_TCP configuration settings are described in Table 4-44.
Table 4-44. Make_Call_TCP Activity Test Settings
Category Settings
Scenario Editor The SC_018_B2B_SIPv4_T2_ReceiveCall_w_RTP_two_Clouds tst test scenario comprises a basic call originating sequence on the first channel, and a call terminating sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
For each media channel, a unique (IP, port) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• UDP port: Use same value
Dial Plan The Receive_Call_TCP activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
The Override transport option is enabled and TCP transport is selected.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints establish signaling and media calls, the Enable media on this activity option is selected.
The RTP port is configured to the same 10000 value for all channels.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: Receive_Call_TCP uses the same settings as Make_Call_TCP, except for the Dial Plan page, which does not need to specify a destination activity.
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The IM_Send_Receive configuration settings are described in Table 4-45.
SC_019_B2B_SIPv6_T1_MakeCall_from_Cloud
This test is similar to SC_001_B2B_SIPv4_T1_MakeCall_from_Cloud, except that the SIP endpoints are configured using IPv6 instead of IPv4 settings.
Table 4-45. IM_Send_Receive Activity Test Settings
Category Settings
Scenario Editor The SC_SC_019_B2B_SIPv4_T2_Instant_Messaging_two_Clouds test scenario comprises a basic instant messaging send and receive sequence on the first channel, and a receive and send sequence on the second channel.
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The IM_Receive_Send activity is configured as destination activity.
SIP Settings The SIP port has the default setting [5060-]; only the first value of the series is used because of the TCP/UDP/TLS port =Use same value setting from the Execution Settings page.
Cloud SIP Settings This activity is assigned to VoIPSIPCloud1, which emulates two SIP Proxy servers, both configured with Use Server (servers are included in the initial messages path) and Keep in Route (servers remain in the subsequent messages path) options selected.
No overriding dispatching rules are defined.
Codec Settings The default codec settings are used.
RTP Settings Since endpoints perform only instant messaging procedures, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note: IM_Receive_Send uses the same settings as IM_Send_Receive, except for the Dial Plan page, which specifies IM_Send_Receive as destination activity.
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SC_020_B2B_SIPv6_T1_MakeCall_from_Cloud_RTP_1_to_n
This test is similar to SC_003_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_ 1_to_n, except that the SIP endpoints are configured using IPv6 instead of IPv4 settings.
SC_021_B2B_SIPv6_T1_ReceiveCall_by_Cloud
This test is similar to SC_005_B2B_SIPv4_T1_ReceiveCall_by_Cloud, except that the SIP endpoints are configured using IPv6 instead of IPv4 settings.
SC_022_B2B_SIPv6_T2_BasicCall_between_two_Clouds
This test is similar to SC_011_B2B_SIPv4_T2_BasicCall_between_two_Clouds, except that the SIP endpoints are configured using IPv6 instead of IPv4 settings.
SC_023_B2B_SIPv6_T2_BasicCall_between_two_Clouds_no_Routes
This test is similar to SC_012_B2B_SIPv4_T2_BasicCall_between_two_Clouds_no_Routes, except that the SIP endpoints are configured using IPv6 instead of IPv4 settings.
SC_024_B2B_SIPv6_T2_BasicCall_between_two_Clouds_RTP_1_to_1
This test is similar to SC_014_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_1_to_1, except that the SIP endpoints are configured using IPv6 instead of IPv4 settings.
SC_025_B2B_SIPv4_T1_MakeCall_from_Cloud_SRTP_1_to_n
This test is similar to SC_003_B2B_SIPv4_T1_MakeCall_from_Cloud_RTP_ 1_to_n, except that the media traffic exchanged between endpoint is encrypted using SRTP.
SC_026_B2B_SIPv4_T2_BasicCall_between_two_Clouds_SRTP_1_to_1
This test is similar to SC_014_B2B_SIPv4_T2_BasicCall_between_two_Clouds_RTP_1_to_1, except that the media traffic exchanged between endpoint is encrypted using SRTP.
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VS_SMS_001_B2B_SIPv4 UE vs. CSCF Complete SMS Flow
This test runs in Back-to-Back mode and comprises of two VoIPSIPPeer activi-ties, each one includes Submit and Deliver and Status-Report procedures (Figure 4-40). This is a basic SMS flow, where the sender of the SMS corresponds with the receiver.
First activity plays the role of UE and the second represents CSCF. UE sends the SMS Submit message to CSCF, CSCF receives the Submit message, send Deliver to the same UE and then send Submit-Report and because of checked Use Status-Report Indicator.
Table 4-46. Table UE vs. CSCF Complete SMS Flow
Category Settings
Scenario VS_SMS_001_B2B_SIPv4 UE vs. CSCF Complete SMS Flow comprises two channels:
Ch#0: SIP SMS Submit Initiate, SIP SMS Deliver Receive, SIP SMS Status-Report Receive
Ch#1: SIP SMS Submit Receive, SIP SMS Deliver Initiate, SIP SMS Status-Report Initiate
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
Dial Plan The UE activity and port 5060 are configures as destination for CSCF activity.
The CSCF activity and port 5060 are configures as destination for UE activity.
SIP Settings The SIP port has the 5060 default setting for all channels.
SMS Use Status Report Request (SM Originator) is set for both UE and CSCF activities.
File name sms_user_data.ixsms is added to be sent.
Other Settings The IP version preference is set to IPv4 and no scenario variables need to be initialized.
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Figure 4-40. VS_SMS_001_B2B_SIPv4 UE vs. CSCF Complete SMS Flow
VS_SMS_002_DUT_SIPv4 UE vs. UE End-to-end SMS Flow
This test runs against DUT (SUT) and comprises of two VoIPSIPPeer activities, first includes the sender of SM (Sender_UE) and the second represents the receiver of SM (Recipient_UE).
In the scenario channels, there are used two procedures: SMS Submit Initiate (Figure 4-41) for Sender_UE, and SMS Deliver Receiver (Figure 4-42) for Recipient_UE.
Table 4-47. UE vs. UE End-to-end SMS flow
Category Settings
Scenario VS_SMS_002_DUT_SIPv4 UE vs. UE End-to-end SMS flow comprises two channels:
Ch#0: SIP SMS Submit Initiate,
Ch#1: SIP SMS Deliver Receive
Execution Settings For each signaling channel, a unique (IP, port, phone) tuple is generated using the following Channel Mapping settings:
• IP: Use Consecutive values (per port)
• TCP/UDP/TLS port: Use same value
• Phone: Use consecutive values (per port)
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Figure 4-41. VS_SMS_002_DUT_SIPv4 UE vs. UE End-to-end SMS flow - SIP SMS Submit Initiate
Figure 4-42. VS_SMS_002_DUT_SIPv4 UE vs. UE End-to-end SMS flow SIP SMS Deliver Receive
Dial Plan The Sender_UE activity and port 5060 are configures as destination for Recipient_UE activity.
The Recipient_UE activity and port 5060 are configures as destination for Sender_UE activity.
Dial Plan The Sender_UE activity and port 5060 are configures as destination for Recipient_UE activity.
The Recipient_UE activity and port 5060 are configures as destination for Sender_UE activity.
SIP Settings The SIP port has the 5060 default setting for all channels.
SMS File name sms_user_data.ixsms is added to be sent.
Other Settings The IP version preference is set to IPv4 and no scenario variables need to be initialized.
Table 4-47. UE vs. UE End-to-end SMS flow
Category Settings
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Skinny Procedures, Sample Test Con-figurations and Test Scenarios
To get you started more easily with Skinny protocol testing, a number of pre-defined Skinny procedures and sample test configuration building up on existing script functions were developed. This section provides a description of these pre-defined IxLoad Voice Plug-in Skinny procedures, available sample test configu-rations (RXFs) and their associated test scenarios.
Sample Test Configurations Naming Convention
The available test configurations adhere to a naming convention that eases identi-fication of a test composition. Acronyms used for denominating configurations and their meaning are listed in Table 4-48:
For example, SK_001_7960_SO_US_5000_Chs_IPv4_Static_Basic_Call_10s denotes a Skinny signaling-only test, emulating User Side equipment, having an objective of 5000 channels, using IPv4 and DHCP assigned addresses, and executing a basic call with a duration of 10 seconds.
Note: Skinny procedures and sample test configuration are available only when installing the Cisco SCCP library, as described in Appendix A of the Getting Started with IxLoad manual.
Note: For a complete description of the available Skinny test library functions, refer to the VoIP Skinny Functions Library on page 3-41.
Table 4-48. Configuration Naming Fields
Field Description
Protocol Skinny, Mixed (SIP and Skinny)
nnn Sequential number (001-999)
Phone type Emulated IP phone type (7902, 7960)
Test type Signaling Only (SO), Signaling and Media (SM)
Network type User Side (US), Network Side (NS), User Side and Network Side (USNS)
Objective value Numerical value
Objective type Channels (Chs), BHCA
IP version IPv4, IPv6
IP Type DHCP, Static
Short Description The name of the sample file
Note: Other possible descriptors, such as HL or LL, can also be used to differentiate between library types, such as Low Level (LL) or High Level (HL).
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Skinny Predefined Procedures
A procedure is a simple way to encapsulate several script functions into a single function block that can be re-used within a number of scenarios.
Based on the functions from the Skinny test library, the Skinny predefined proce-dures described in Table 4-49 (available in the Procedure Library) were devel-oped.
Table 4-49. Skinny Predefined Procedures
Category Procedure Description
High Level SK_IsOnHook This procedure retrieves the values of the Skinny scenario variables and uses a VariableTest script function to check if the call state is ‘onhook’ ($Sk_CallState=2).
SK_IsOffHook This procedure retrieves the values of the Skinny scenario variables and uses a VariableTest script function to check if the call state is ‘onhook’ ($Sk_CallState=1).
SK_IsRingIn This procedure retrieves the values of the Skinny scenario variables and uses a VariableTest script function to check if the call state is ‘onhook’ ($Sk_CallState=4).
SK_IsRingOut This procedure retrieves the values of the Skinny scenario variables and uses a VariableTest script function to check if the call state is ‘onhook’ ($Sk_CallState=3).
SK_IsConnected This procedure retrieves the values of the Skinny scenario variables and uses a VariableTest script function to check if the call state is ‘onhook’ ($Sk_CallState=5).
SK_IsHold This procedure retrieves the values of the Skinny scenario variables and uses a VariableTest script function to check if the call state is ‘hold’ ($Sk_CallState=8).
SK_REG_7902 This procedure registers a 7902 type phone with a CallManager.
SK_REG_7910 This procedure registers a 7910 type phone with a CallManager.
SK_REG_7940 This procedure registers a 7940 type phone with a CallManager.
SK_REG_7960 This procedure registers a 7960 type phone with a CallManager.
SK_DeREG This procedure de-registers a phone.
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SK_Receive and AnswerCall_HL
This procedure, comprising a Wait Call and an Answer Call script function, is the correspondent for the receiving channel of a Make Call function.
Low Level SK_Empty Message Queue_LL
This procedure empties the Skinny messages queue.
SK_Make Call LL This procedure originates a call by dialing a dial-plan specified number and waiting for a Connected message. It uses Low Level Skinny functions, softkeys and events for initiating a call.
SK_Receive Call_LL This procedures uses Low Level Skinny functions, softkeys and events for receiving a call.
SK_Make Call to BL LL
The procedure originates a call by dialing a Skinny dial-plan specified number and waits for a StartToneMessage with a DtAlertingTone value.
SK_Hold_LL This procedures uses Low Level Skinny functions, softkeys and events for putting a call on hold.
SK_Retrieve_LL This procedures uses Low Level Skinny functions, softkeys and events for retrieving a call from the hold state.
SK_End Call Initiate_LL
This procedures for uses Low Level Skinny functions, softkeys and events to initiate a call closure.
SK_End Call Terminate_LL
This procedures uses Low Level Skinny functions, softkeys and events to respond to a call closure.
Table 4-49. Skinny Predefined Procedures (Continued)
Category Procedure Description
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Skinny Test Configurations
This section provides a brief description of the sample VoIPSkinny Peer test con-figuration files that can be installed and used with the IxLoad Voice Plug-in.
Skinny tests fall into either of the following categories:
• Skinny Signaling Only
• Skinny Bulk Calls
• Advanced Call Features
• Mixed Skinny and SIP - SIP UAs
• Mixed Skinny and SIP - SIP Trunk
Skinny Signaling Only
Tests in this category execute Skinny registration and de-registration procedures with a specified Cisco CallManager. Tests comprise a single activity configured to emulate a number of 100 phones that execute the registration and de-registra-tion operations repeatedly during the test sustain time.
SK_001_7902_SO_US_100_Chs_IPv4_Static_Seq_Registration_5_retries
This signaling-only test illustrates a phone registration procedure, whereby a number of 7902 type phones emulated by the SK_SEP7902A0000001 activity attempt registration with a specified Cisco CallManager.
This registration procedure is executed repeatedly for the entire sustain time duration.
The underlying one-channel test scenario comprising the functions flow executed by the SK_SEP7902A0000001 emulated phones is shown in Figure 4-43.
Figure 4-43. SK_001_7902_SO_US_100_Chs_IPv4_Static_Seq_Registration_5_retries Test Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-50.
Note: An overview table comprising the most important configured parameters for all Skinny sample test configuration is provided in Chapter E, Skinny Sample Configurations Overview.
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SK_002_7960_SO_US_100_Chs_IPv4_Static_Seq_Registration_5_retries
This signaling-only test illustrates a phone registration procedure, with the differ-ence that the emulated Skinny phones are of the 7960 type.
SK_005_7902_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop
This signaling-only test illustrates a phone registration procedure, whereby a number of 7902 type phones emulated by the SK_SEP7902A0000001 activity continuously attempt registration followed by de-registration with the Cisco CallManager. Registration is followed by de-registration.
This registration and de-registration procedure is executed repeatedly for the entire sustain time duration.
Table 4-50. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario is completely configured and comprises the predefined Skinny SK_REG_7902 procedure. Within the procedure, phone A on the first scenario channel executes a registration with the CallManager specified in the Skinny Settings page.
Registration is attempted successively for five times (the RetriesNo variable is initialized to a value of ‘5’). If the registration cannot be completed successfully after the fifth attempt, the procedure exits on the Error output.
Execution Settings The corresponding scenario channel is configured to execute 1 loop during sustain time and to use 1 alias/channel.
Dial Plan The phone registration names are defined using a SEP7902A00[00001-] sequence generating expression that generates 100 registration name strings.
Call and transfer destination need not configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The codec settings can be left unchanged.
RTP Settings Since the activity does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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The underlying one-channel test scenario is shown in Figure 4-44.
Figure 4-44. SK_005_7902_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop Test Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-51.
SK_006_7960_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop
This signaling-only test illustrates a phone registration procedure similar to the previous test, with the only difference that the emulated Skinny IP phones are of the 7960 type.
Table 4-51. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario is completely configured and comprises the Register Client and Unregister Client script functions.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the sustain time and to use 1 alias/channel.
Dial Plan The phone registration names are defined using a SEP7902A00[00001-] sequence generating expression that generates 100 registration name strings.
Call and transfer destination need not configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The codec settings can be left unchanged.
RTP Settings Since the activity does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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SK_003_7902_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop_5_retries
This signaling-only test illustrates a phone registration procedure, whereby a number of 7902 type phones emulated by the SK_SEP7902A0000001 activity continuously attempt registration followed by de-registration with the Cisco Call-Manager.
This registration and de-registration procedure executes repeatedly for the entire test sustain time duration.
The underlying one-channel test scenario is shown in Figure 4-45.
Figure 4-45. SK_003_7902_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop_5_retries Test Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-52.
Table 4-52. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario channel comprises the Sk_REG_7902 procedure followed by a Skinny Unregister Client script function.
Using the Sk_REG_7902 procedure registration is attempted successively for five times (the RetriesNo variable is initialized to a value of ‘5’). If the registration cannot be completed successfully after the fifth attempt, the procedure exits on the Error output.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the sustain time and to use 1 alias/channel.
Dial Plan The phone registration names are defined using a SEP7902A00[00001-] sequence generating expression that generates 100 registration name strings.
Call and transfer destination need not configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The codec settings can be left unchanged.
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SK_004_7960_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop_5_retries
This signaling-only test illustrates a phone registration procedure similar to that of the previous test, with the difference that the emulated Skinny phones are of the 7960 type.
SK_007_7902_SO_US_100_Chs_IPv4_Static_Seq_Reg_5s_Sleep_Dereg_5_retries
This signaling-only test illustrates a phone registration procedure, whereby a number of 7902 type phones emulated by the SK_SEP7902A0000001 activity attempt de-registration with the Cisco CallManager. After being registered for a 5 seconds duration, the phones de-register.
This registration and de-registration sequence executes repeatedly for the entire sustain time duration.
The underlying one-channel test scenario is shown in Figure 4-46.
Figure 4-46. SK_004_7960_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop_5_retries
SK_SEP7902A0000001 configured settings are described in Table 4-53.
RTP Settings Since the activity does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-52. SK_SEP7902A0000001 Activity Test Settings
Category Settings
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SK_008_7960_SO_US_100_Chs_IPv4_Static_Seq_Reg_5s_sleep_Dereg_5_retries
This signaling-only test illustrates a phone de-registration procedure similar to that of the previous test, with the difference that the emulated Skinny phones are of the 7960 type.
Table 4-53. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario channel comprises the Skinny Register Client function, followed by a Sleep function for maintaining registration for a 5 seconds duration. Phones de-registrations finally occurs using the Skinny DeREG procedure.
Using the DeREG procedure de-registration is attempted successively for five times (the RetriesNo variable is initialized to a value of ‘5’). If the de-registration cannot be completed successfully after the fifth attempt, the procedure exits on the Error output.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the sustain time and to use 1 alias/channel.
Dial Plan The phone registration names are defined using a SEP7902A00[00001-] sequence generating expression that generates 100 registration name strings.
Call and transfer destination need not configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The codec settings can be left unchanged.
RTP Settings Since the activity does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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Skinny Bulk Calls
This test samples group is aimed at stressing the Cisco CallManager with bulk calls, both signaling-only and signaling with media streaming, between emulated Cisco 7902 IP phones.
Activities in this test group are configured to emulate large number of phones and to execute repeatedly during the test sustain time. Both signaling-only and signaling with media streaming test samples are available.
SK_009_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_10s
This signaling-only test illustrates a basic call procedure, whereby a number of 7902-type Skinny phones emulated by the SK_SEP7902A0000001 activity estab-lish calls with the Skinny phones emulated by the SK_SEP7902B0000001 activ-ity. After establishing a call, the phones remain in an active call state for the duration of the Sleep script function before the call is terminated.
This call procedure is executed repeatedly for the entire test sustain time.
The underlying two-channel test scenario involving Skinny phones A and B is shown in Figure 4-47.
Figure 4-47. SK_009_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_10sTest Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-54.
Phone A
Phone B
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Table 4-54. VoIPSkinny Peer1 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises phone registrations with the Cisco CallManager followed by a call establishment procedure. Registration is performed during first loop only.
Phone A of the first scenario channel uses a Skinny Make Call script function to initiate a call based on the dial plan settings.
The Sk_Receive AnswerCall_HL procedure used by phone B of the second channel includes the common Skinny Wait Call and Skinny Answer Call script functions. After establishing the call, Sleep function with a 10 seconds duration is executed by both phone A and phone B.
Eventually the call is terminated by phone A using a Skinny End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time and to use 1 alias/channel.
Dial Plan The phone registration names are defined using a SEP7902A00[00001-] sequence generating expression.
SK_SEP7902B0000001 is configured as a call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The codec settings are of no importance and can be left unchanged, since no RTP streaming is performed by the test.
RTP Settings Since the activity does not perform any RTP streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7902B0000001 uses the same settings as SK_SEP7902A0000001, except for the Dial Plan page, which does not specify any call or call transfer destination. The registration names of the SK_SEP7902B0000001 - emulated phones are specified using a SEP7902B00[00001-] sequence generating expression.
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SK_010_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_3min
This test is similar to the previous one, with the only difference that the call dura-tion is 3 minutes.
SK_011_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_30min
This test is similar to the previous one, with the only difference that the call dura-tion is 30 minutes.
SK_012_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_10s
This test illustrates a basic call procedure with media streaming, whereby a num-ber of 7902 phones emulated by the SK_SEP7902A0000001 activity establish calls with the Skinny phones emulated by the SK_SEP7902B0000001 activity. After establishing a call, the phones perform bidirectional media streaming before the call is terminated.
This call procedure is repeated for the entire test sustain time.
The underlying two channel test scenario involving Skinny phones A and B is shown in Figure 4-48.
Figure 4-48. SK_012_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_10s Test Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-55.
Phone A
Phone B
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Table 4-55. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario channel comprises the phones registration with the CallManager followed by a call establishment procedure.
Phone A of the first scenario channel uses a Skinny Make Call script function to initiate a call based on the dial plan settings. After establishing the call, media streaming is performed using Voice Session script functions on both scenario channels. Finally the call is terminated by phone A using a Skinny End Call procedure.
The Sk_Receive AnswerCall_HL procedure used by phone B on the second channel is a wrapping of the common Skinny Wait Call and Skinny Answer Call script functions.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7902A00[00001-] sequence generating expression.
SK_SEP7902B0000001 is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The codec settings can be left unchanged.
RTP Settings The Enable media on this activity option is selected for media streaming to be performed between the emulated Skinny phones.
Other Settings The IP version preference is set to IPv4, and no scenario variables need initialized.
Note:
SK_SEP7902B0000001 uses the same settings as SK_SEP7902A0000001, except for the Dial Plan page, which does not specify any call or call transfer destination. The registration names of the SK_SEP7902B0000001-emulated phones are specified using a SEP7902B00[00001-] sequence generating expression.
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SK_013_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_3min
This test is similar to the previous one, with the only difference that the call dura-tion is 3 minutes. The longer call duration is obtained by playing the selected wave file continuously for 3 minutes (Advanced Playback Settings tab of the Voice Session script function).
SK_014_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_30min
This test is similar to the previous one, with the only difference that the call dura-tion is 30 minutes. The longer call duration is obtained by playing the selected wave file continuously for 3 minutes (Advanced Playback Settings tab of the Voice Session script function).
SK_016_7902_SM_US_300_Chs_IPv4_Static_Basic_Call_DTMFs_inband_3min_7902
This test illustrates a basic call procedure with inband transmission of DTMFs. A number of 7902 phones emulated by the SK_SEP7902A0000001 activity estab-lish calls with the phones emulated by the SK_SEP7902B0000001 activity. After the call is established, the phones use the Generate DTMF / Detect DTMF script functions for bidirectional transmission and detection of DTMFs.
This call procedure is repeated for the entire test sustain time.
The underlying two-channel test scenario involving phones A and B is shown in Figure 4-49.
Figure 4-49. SK_016_7902_SM_US_300_Chs_IPv4_Static_Basic_Call_DTMFs_inband_3min_7902 Test Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-56.
Phone A
Phone B
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Table 4-56. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure.
Phone A on the first scenario channel uses a Skinny Make Call script function to initiate a call based on the dial plan settings.
The Sk_Receive AnswerCall_HL procedure used by phone B on the second channel is a wrapping of the common Skinny Wait Call and Skinny Answer Call script functions.
After establishing the call, DTMFs are transmitted bidirectionally inband using Generate DTMF script functions having the inband transmission mode selected. DTMFs are detected using Detect DTMF functions.
Eventually the call is terminated by phone A using a Skinny End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960A00[00001-] sequence generating expression.
SK_SEP7902B0000001 is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The G.711 codec is selected for the DTMFs to be transmitted inband.
RTP Settings The Enable media on this activity option is selected for the DTMFs to be transmitted inband.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7902B0000001 uses the same settings as SK_SEP7902A0000001, except for the Dial Plan page, which does not specify any call or call transfer destination. The registration names of the SK_SEP7902B0000001 - emulated phones are specified using a SEP7960B00[00001-] sequence generating expression.
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SK_017_7902_SM_US_300_Chs_IPv4_Static_Basic_Call_DTMFs_out-of-band_3min_7902
This test is similar to the previous one, with the difference that the DTMFs are transmitted out-of-band using the RTP 2833 Event payload format in the Generate DTMF script functions.
SK_021_7902_SM_US_300_Chs_IPv4_Static_BasicCall_Tone_highFreq_inband_3min
This test illustrates a basic call procedure with inband transmission and detection of custom tones. A number of 7902 phones emulated by the SK_SEP7902A0000001 activity establish calls with the phones emulated by the SK_SEP7902B0000001 activity. After establishing the call, the phones use the Generate Tone / Wait for Tone script functions for transmitting and detecting custom tones.
This call procedure is repeated for the entire test sustain time.
The underlying two-channel test scenario involving phones A and B is shown in Figure 4-50.
Figure 4-50. SK_021_7902_SM_US_300_Chs_IPv4_Static_BasicCall_Tone_highFreq_inband_3min Test Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-57.
Phone A
Phone B
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Table 4-57. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure.
Phone A on the first scenario channel uses a Skinny Make Call script function to initiate a call based on the Dial Plan settings.
The Sk_Receive AnswerCall_HL procedure used by phone B on the second channel is a wrapping of the common Skinny Wait Call and Skinny Answer Call script functions.
After establishing the call, custom tones are transmitted bidirectionally inband using Generate Tone script functions having the inband transmission mode selected. On the receiving sides, tones are detected using Wait for Tone functions.
Eventually the call is terminated by phone A using a Skinny End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7902A00[00001-] sequence generating expression.
SK_SEP7902B0000001 is configured as call destination, no transfer destination needs configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The G.711 codec is selected for the tone to be transmitted inband.
RTP Settings The Enable media on this activity option is selected for the tone to be transmitted inband.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7902B0000001 uses the same settings as SK_SEP7902A0000001, except for the Dial Plan page, which does not specify any call or call transfer destination. The registration names of the SK_SEP7902B0000001-emulated phones are specified using a SEP7902B00[00001-] sequence generating expression.
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SK_020_7902_SM_US_300_Chs_IPv4_Static_BasicCall_Tone_medFreq_inband_3min
The test is the same as the previous one, with the difference that a different cus-tom tone is transmitted by the Generate Tone functions.
SK_019_7902_SM_US_300_Chs_IPv4_Static_BasicCall_Tone_lowFreq_inband_3min
The test is the same as the previous one, with the difference that a different cus-tom tone is transmitted by the Generate Tone functions.
SK_015_7902_SM_US_10K_BHCA_IPv4_Static_Basic_Call_Voice_1min
This test illustrates a basic call procedure with media streaming, having a config-ured BHCA test objective of 10000 calls/hour. A number of 7902 type phones emulated by the SK_SEP7902A0000001 activity establish calls with the phones emulated by the SK_SEP7902B0000001 activity. After establishing the call, the phones use the Voice Session functions for bidirectional media streaming.
This call procedure is executed repeatedly for the entire test sustain time.
The underlying two-channel test scenario involving phones A and B is shown in Figure 4-51.
Figure 4-51. SK_015_7902_SM_US_10K_BHCA_IPv4_Static_Basic_Call_Voice_1min Test Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-58.
Note: The BHCA objective of 10000 calls/hour is configured with a talk time of 40 seconds, slightly longer than the duration of the wave file played by the Voice Session functions on each scenario channel.
Phone A
Phone B
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Table 4-58. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration with the Call Manager using the Dial Plan settings, followed by a call establishment procedure.
Phone A on the first scenario channel uses a Skinny Make Call script function to initiate a call based on the Dial Plan settings.
The Sk_Receive AnswerCall_HL procedure used by phone B on the second channel is a wrapping of the common Skinny Wait Call and Skinny Answer Call script functions. The Voice Session functions used on both channels for bidirectional media streaming each play a wave file with a duration of about 33 seconds.
Eventually the call is terminated by phone A using a Skinny End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7902A00[00001-] sequence generating expression.
SK_SEP7902B0000001 is configured as call destination, no transfer destination needs configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings Default codec settings are used.
RTP Settings The Enable media on this activity option is selected for media streaming to be performed between the emulated phones.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7902B0000001 uses the same settings as SK_SEP7902A0000001, except for the Dial Plan page, which does not specify any call or call transfer destination. The registration names of the SK_SEP7902B0000001 - emulated phones are specified using a SEP7902B00[00001-] sequence generating expression.
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SK_018_7902_SM_US_25K_BHCA_IPv4_Static_Basic_Call_DTMFs_inband_3_min
This test illustrates a basic call procedure with inband transmission of custom tones, having a configured BHCA test objective of 25000 calls/hour. The 7902 phones emulated by the SK_SEP7902A0000001 activity establish calls with the phones emulated by the SK_SEP7902B0000001 activity. After establishing the call, the phones use the Generate Tone / Wait for Tone functions for generating and detecting custom tones.
This call procedure is repeated for the entire test sustain time.
The underlying two-channel test scenario phones A and B is shown in Figure 4-52.
Figure 4-52. SK_018_7902_SM_US_25K_BHCA_IPv4_Static_Basic_Call_DTMFs_inband_3_min Test Scenario
SK_SEP7902A0000001 configured settings are described in Table 4-59.
Note: The BHCA objective of 25000 calls/hour is configured with a talk time of approximately 120 seconds.
Phone A
Phone B
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Table 4-59. SK_SEP7902A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure. After establishing the call, DTMFs are transmitted bidirectionally using the Generate Tone / Wait for Tone script functions on both scenario channels.
Phone A on the first scenario channel uses a Skinny Make Call script function to initiate a call based on the Dial Plan settings.
The Sk_Receive AnswerCall_HL procedure used by the second scenario channel is a wrapping of the common Skinny Wait Call and Skinny Answer Call script functions.
Eventually the call is terminated by phone A using a Skinny End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7902A00[00001-] sequence generating expression.
SK_SEP7902B0000001 is configured as call destination, no transfer destination needs configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The Sequential registration and the Fail if previously has failed options are selected.
Codec Settings The G.711 codec is selected for the tone to be transmitted inband.
RTP Settings The Enable media on this activity option is selected for media streaming to be performed between the emulated phones.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7902B0000001 uses the same settings as SK_SEP7902A0000001, except for the Dial Plan page, which does not specify any call or call transfer destination. The registration names of the SK_SEP7902B0000001 - emulated phones are specified using a SEP7902B00[00001-] sequence generating expression.
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Advanced Call Features
This tests group is aimed at testing advanced call features provided by the Cisco CallManager, such as forward, conference calls, or hold/retrieve.
VoIPSkinnyPeer activities in this group are configured to emulate a number of five 7960-type Skinny phones, and tests execute a single loop during test sustain time.
SK_022_7960_SM_US_5_Chs_IPv4_Static_Hold_Resume
This test, illustrating a call hold and retrieve procedure, comprises two VoIP-Skinny Peer activities: phone A emulated by SK_SEP7960AF000001 establishes a call with the phone B emulated SK_SEP7960BF000001, then puts the call on hold and remains idle for the duration of the Sleep script function.
Eventually phone A retrieves the call to B previously put on hold and performs another voice session (Figure 4-53).
Figure 4-53. SK_022_7960_SM_US_5_Chs_IPv4_Static_Hold_Resume Test Scenario
SK_SEP7960AF000001 configured settings are described in Table 4-60.
Note: The network settings for all tests described subsequently are the same, namely 1 IPv4 host with no emulated router.
Phone A
Phone B
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SK_031_7960_SM_US_5_Chs_IPv4_Static_List_Ad_Hoc_Conference
This test, illustrating a call conference procedure, comprises three VoIPSkinny Peer activities: phone A emulated by SK_SEP7960AF000001 establishes a call with the phone B emulated SK_SEP7960BF000001 and performs media stream-ing. Using the SetUp Xfer and the Complete Xfer script functions, phone A establishes a conference call with phone C of SK_SEP7960CF000001, joining all phones in a conference.
Table 4-60. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor The two channels test scenario is completely configured.
Using Dial Plan settings, phone A on the first scenario channel establishes a call with phone B on the second channel. After putting the call on hold, phone A remains idle for the duration of the Sleep function and then retrieves the call.
Media streaming using the Voice Session script function is performed each time two phones get connected.
Execution Settings The corresponding scenario channel is configured to execute once during test sustain time and to use 1 alias/voice channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
SK_SEP7960BF000001 is configured as a call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The codec settings can be left unchanged.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call or call transfer destination. The registration names are defined using a SEP7960BF0[00001-] sequence generating expression.
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Once connected in an Ad-Hoc conference, all phones perform media streaming. Eventually phone A presses the SkConfList softkey displaying all parties involved in the conference (Figure 4-54).
Figure 4-54. SK_031_7960_SM_US_5_Chs_IPv4_Static_List_Ad_Hoc_Conference Test Scenario
SK_SEP7960AF000001 configured settings are described in Table 4-61.
Phone A
Phone B
Phone C
Table 4-61. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor Using Dial Plan settings, phone A on the first scenario channel establishes a call with phone B on the second channel. After performing a media session, phone A establishes a conference call with phone C and joins all calls using a Complete Xfer script function.
Eventually phone A presses the SkConfList softkey displaying all parties involved in the conference call.
Execution Settings The corresponding scenario channel is configured to execute once during test sustain time and to use 1 alias/voice channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
SK_SEP7960BF000001 is configured as a call destination, while SK_SEP7960CF000001 is configured as call transfer destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The codec settings can be left unchanged.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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SK_032_7960_SM_US_5_Chs_IPv4_Static_Forward_All_Calls
This test, illustrating a call forwarding procedure, comprises three VoIPSkinny Peer activities: phone A emulated by SK_SEP7960AF000001 originates a call to SK_SEP7960BF000001, while phone B emulated by SK_SEP7960BF000001 executes a Skinny ForwardAllCalls function, instructing the CallManager to forward the call to phone C emulated by SK_SEP7960CF000001.
After the call is established between phones A and C, media streaming is per-formed between the two parties (Figure 4-55).
Figure 4-55. SK_032_7960_SM_US_5_Chs_IPv4_Static_Forward_All_Calls
VoIPSkinnyPeer1 configured settings are described in Table 4-62.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960BF0[00001-] sequence generating expression.
SK_SEP7960CF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960CF0[00001-] sequence generating expression.
Phone A
Phone B
Phone C
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SK_033_7960_SM_US_5_Chs_IPv4_Static_Forward_Busy
This test, illustrating a call forwarding procedure, comprises four VoIPSkinny Peer activities: after phone A emulated by SK_SEP7960AF400001 has estab-lished a call with phone B emulated by SK_SEP7960BF400001, phone C corre-sponding to SK_SEP7960CF400001 calls phone B. Since phone B is already involved in a call, configured CallManager settings determine the call to be for-
Table 4-62. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor The three channels test scenario is completely configured.
Phone A on the first scenario channel phone attempts to establish a call with the phone B, which executes a Skinny ForwardAllCalls script function and thus instructs the CallManager to forward the call to phone C on the third channel.
The Sk_Receive AnswerCall_HL procedure used by phone C to answer the incoming call is a wrapping of the common Skinny Wait Call and Skinny Answer Call script functions.
Execution Settings The corresponding scenario channel is configured to execute once during test sustain time and to use 1 alias/voice channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
The SK_SEP7960BF000001 activity is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The default codec settings are used.
RTP Settings Since this activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify a call destination. Its registration names are defined using a SEP7960BF0[00001-] sequence generating expression.
SK_SEP7960CF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not need to specify a call destination activity, since it only terminates a call. Its registration names are defined using a SEP7960CF0[00001-] sequence generating expression.
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warded to phone D corresponding to SK_SEP7960DF400001, which answers the call (Figure 4-56).
Figure 4-56. SK_033_7960_SM_US_5_Chs_IPv4_Static_Forward_Busy Test Scenario
SK_SEP7960AF400001 configured settings are described in Table 4-63.
Phone A
Phone B
Phone C
Phone D
Table 4-63. SK_SEP7960AF400001 Activity Test Settings
Category Settings
Scenario Editor The four channels test scenario is completely configured.
Execution Settings The corresponding scenario channel is configured to execute once during test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF4[00001-] sequence generating expression.
SK_SEP7960BF400001 is configured as a call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The codec settings can be left unchanged.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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SK_034_7960_SM_US_5_Chs_IPv4_Static_Forward_No_Answer
This test, illustrating a call forward procedure, comprises three VoIPSkinny Peer activities: phone A emulated by SK_SEP7960AF500001 calls phone B emulated by SK_SEP7960BF500001, which forwards the call to phone C corresponding to SK_SEP7960CF500001 (Figure 4-57).
Figure 4-57. SK_034_7960_SM_US_5_Chs_IPv4_Static_Forward_No_Answer Test Scenario
SK_SEP7960AF500001 configured settings are described in Table 4-64.
Note:
SK_SEP7960BF400001 uses the same settings as SK_SEP7960AF400001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960BF4[00001-] sequence generating expression.
SK_SEP7960CF400001 uses the same settings as SK_SEP7960AF400001, except for the Dial Plan page, which specifies SK_SEP7960BF400001 as call destination, since it attempts to establish a call with it. Its registration names are defined using a SEP7960CF4[00001-] sequence generating expression.
SK_SEP7960DF400001, the final destination of the call initiated by SK_SEP7960CF400001, is configured the same as SK_SEP7960AF400001, except for the Dial Plan page, which does not need to specify any call or call transfer destination. Its registration names are defined using a SEP7960DF4[00001-] sequence generating expression.
Important:
In order for this test to execute correctly, call forwarding on a busy status for the Skinny phone B emulated by SK_SEP7960BF400001 needs to be configured on the Cisco CallManager.
Phone A
Phone B
Phone C
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Table 4-64. SK_SEP7960AF500001 Activity Test Settings
Category Settings
Scenario Editor The three channels test scenario is completely configured.
Using Dial Plan settings, phone A of SK_SEP7960AF500001 calls phone B of SK_SEP7960BF500001, which does not answer the call and forwards the call to phone C of SK_SEP7960CF500001.
Execution Settings The corresponding scenario channel is configured to execute once during test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF5[00001-] sequence generating expression.
SK_SEP7960BF500001 is configured as a call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The codec settings can be left unchanged.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF500001 uses the same settings as SK_SEP7960AF500001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960BF5[00001-] sequence generating expression.
SK_SEP7960CF500001 uses the same settings as SK_SEP7960AF500001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960CF5[00001-] sequence generating expression.
Important:
In order for this test to execute correctly, call forwarding on a no answer status for the Skinny phone emulated by SK_SEP7960BF500001 needs to be configured on the Cisco CallManager.
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SK_026_7960_SM_US_5_Chs_IPv4_Static_Ad_hoc_Conference
This test, illustrating an Adhoc conference procedure, comprises three VoIP-Skinny Peer activities: phone A emulated by SK_SEP7960AF000001 originates a call to phone B emulated by SK_SEP7960BF000001, then creates a conference call to phone C of SK_SEP7960CF000001 and finally joins all parties in an Ad-hoc conference.
After the conference is established, all parties perform voice sessions between them (Figure 4-58).
Figure 4-58. SK_026_7960_SM_US_5_Chs_IPv4_Static_Ad_hoc_Conference Test Scenario
SK_SEP7960AF000001 configured settings are described in Table 4-65.
Phone A
Phone B
Phone C
Table 4-65. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario is completely configured.
Execution Settings The corresponding scenario channel is configured to execute once during test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
The SK_SEP7960BF000001 activity is configured as call destination, SK_SEP7960CF000001 is configured as call transfer / conference destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The codec settings can be left unchanged.
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SK_027_7960_SM_US_5_Chs_IPv4_Static_MeetMe_Conference
This test, illustrating a MeetMe conference procedure, comprises three VoIP-Skinny Peer activities: phone A emulated by SK_SEP7960AF000001 establishes a call to a specified conference number, while phones B and C, corres-ponding to SK_SEP7960BF000001 and SK_SEP7960CF000001 respectively, join the con-ference by also establishing calls to that number.
After the conference connection is established, all parties perform voice sessions (Figure 4-59).
Figure 4-59. SK_027_7960_SM_US_5_Chs_IPv4_Static_MeetMe_Conference Test Scenario
SK_SEP7960AF000001 configured settings are described in Table 4-66.
RTP Settings Since this activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960BF0[00001-] sequence generating expression.
SK_SEP7960CF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does specify any call or call transfer destination. Its registration names are defined using a SEP7960CF0[00001-] sequence generating expression.
Table 4-65. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Phone A
Phone B
Phone C
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SK_028_7960_SM_US_5_Chs_IPv4_Static_Join_2_Calls
This test, illustrating a call joining procedure, comprises three VoIPSkinnyPeer activities: phone A emulated by SK_SEP7960AF000001 establishes a call with phone B emulated by SK_SEP7960BF000001, while phone B and phone C of SK_SEP7960CF000001 also connect in a call. After phone B presses the SkJoin softkey, the calls involving the 3 phones are joined and all parties perform voice sessions (Figure 4-60).
Table 4-66. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor The three channel test scenario is completely configured.
The Skinny Meet-Me script function used by phone A on the first scenario channel sets up the conference by dialing to the pre-configured number (‘6000’ in the case of this sample test), while the Skinny Make Call functions on the other channels dial to the same number to join the conference.
Execution Settings The corresponding scenario channel is configured to execute once during test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
Since the MeetMe conference number is specified as a string in the MeetMe script function, no activity is configured as either a call destination, or a call transfer/conference destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings Default codec settings are used.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for its registration names, defined using a SEP7960BF0[00001-] sequence generating expression.
SK_SEP7960CF000001 uses the same settings as SK_SEP7960AF000001, except for its registration names, defined using a SEP7960CF0[00001-] sequence generating expression.
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Figure 4-60. SK_028_7960_SM_US_5_Chs_IPv4_Static_Join_2_Calls Test Scenario
SK_SEP7960AF000001 configured settings are described in Table 4-67.
Phone A
Phone B
Phone C
Table 4-67. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor The three channel test scenario is completely configured.
After registration of the phones A, B, and C with the CallManager specified by the Dial Plan settings, phone A connects to phone B, while phone C connects to phone B.
Phone B executes the Skinny Send Softkey function with the SkJoin value that joins the calls.
Execution Settings The corresponding scenario channel is configured to execute once during test sustain time and to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
SK_SEP7960BF000001 is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The default codec settings are used.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
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SK_024_7960_SM_US_5_Chs_IPv4_Static_Blind_Transfer
This test, illustrating a blind transfer procedure, comprises three VoIPSkinny Peer activities: phone A emulated by SK_SEP7960AF000001 establishes a call to phone B emulated by SK_SEP7960BF000001 and then transfers the call directly to phone C of SK_SEP7960CF000001.
After the call transfer occurs, phone A goes on hook, and the two parties involved in the call, phone B and phone C, perform voice sessions (Figure 4-61).
Figure 4-61. SK_024_7960_SM_US_5_Chs_IPv4_Static_Blind_Transfer Test Scenario
SK_SEP7960AF000001 configured settings are described in Table 4-68.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does specify any call destination or call transfer destination. Its registration names are defined using a SEP7960BF0[00001-] sequence generating expression.
SK_SEP7960CF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which specifies SK_SEP7960BF000001 as call destination. Its registration names are defined using a SEP7960CF0[00001-] sequence generating expression.
Phone A
Phone B
Phone C
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Table 4-68. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor The three channel test scenario channel is completely configured.
Phone A on the first scenario channel phone calls phone B using a Skinny Make Call script function, while B answers the call using the common Skinny Wait Call and Skinny Answer Call script functions.
After the execution of a voice session between phones A and B, phone A transfers the call to phone C by executing a Skinny Transfer script function with the Blind Transfer option selected. Once connected, phones B and C perform media streaming.
Execution Settings The corresponding scenario channel is configured to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
SK_SEP7960BF000001 is configured as a call destination, while SK_SEP7960CF000001 is configured as a call transfer/conference destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The default codec settings are used.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960BF0[00001-] sequence generating expression.
SK_SEP7960CF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call destination or call transfer destination. Its registration names are defined using a SEP7960CF0[00001-] sequence generating expression.
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SK_023_7960_SM_US_5_Chs_IPv4_Static_Transfer_with_Consultation
This test, illustrating a transfer procedure with consultation, comprises three VoIPSkinny Peer activities: phone A emulated by SK_SEP7960AF000001 estab-lishes a call to phone B emulated by SK_SEP7960BF000001 and then transfers the call to phone C of SK_SEP7960CF000001. The only difference to the previ-ously described test is that phone A performs a voice session (consultation) with phone C prior to transferring the call.
After the call transfer occurs, phone A goes on hook, and the two parties involved in the call, phones B and C, perform voice sessions (Figure 4-62).
Figure 4-62. SK_023_7960_SM_US_5_Chs_IPv4_Static_Transfer_with_Consultation Test Scenario
SK_SEP7960AF000001 configured settings are described in Table 4-69.
Phone A
Phone B
Phone C
Table 4-69. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor The three channel test scenario is completely configured.
Phone A on the first scenario channel phone calls phone B using a Skinny Make Call script function, while B answers the call using the common Skinny Wait Call and Skinny Answer Call script functions.
After the execution of a voice session between phones A and B, phone A initiates the transfer the call to phone C by executing a Skinny Setup XFer followed by a Skinny Complete XFer script function. Once connected, phones B and C perform media streaming.
Execution Settings The corresponding scenario channel is configured to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
SK_SEP7960BF000001 is configured as a call destination, while SK_SEP7960CF000001 is configured as a call transfer/conference destination.
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SK_025_7960_SM_US_5_Chs_IPv4_Static_Direct_Transfer_of_2_parties_on_a_line
This test, illustrating a direct transfer procedure, comprises three VoIPSkinny Peer activities: phone A emulated by SK_SEP7960AF000001 establishes a call to phone B emulated by SK_SEP7960BF000001. Phone C of SK_SEP7960CF000001 also establishes a call with phone B. Phone B then trans-fers the call to phone A.
After the call transfer occurs, phone B goes on hook, and the two parties involved in the call, phones A and C, perform voice sessions (Figure 4-63).
Figure 4-63. SK_025_7960_SM_US_5_Chs_IPv4_Static_Direct_Transfer_of_2_parties_on_a_line Test Scenario
SK_SEP7960AF000001 configured settings are described in Table 4-70.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The default codec settings are used.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960BF0[00001-] sequence generating expression.
SK_SEP7960CF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960CF0[00001-] sequence generating expression.
Table 4-69. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Phone A
Phone B
Phone C
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SK_036_7960_SM_US_5_Chs_IPv4_Static_Call_Group_Pickup
This test, illustrating a call pickup procedure, comprises three VoIPSkinny Peer activities: phone A emulated by SK_SEP7960AF100001 establishes a call to phone B emulated by SK_SEP7960BF100001, which does not answer the call.
Table 4-70. SK_SEP7960AF000001 Activity Test Settings
Category Settings
Scenario Editor The three channel test scenario is completely configured.
Phone A on the first scenario channel calls phone B using a Skinny Make Call script function, and phone B answers the call using the common Skinny Wait Call and Skinny Answer Call script functions. Once connected, the phones perform media streaming.
Phone C also calls phone B which answer the call, media streaming is performed between the connected phones. Phone B then transfers the call to phone A using a SendSoftkey (SkDirTrfr) script function.
Once connected, phones A and C perform media streaming.
Execution Settings The corresponding scenario channel is configured to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF0[00001-] sequence generating expression.
SK_SEP7960BF000001 is configured as a call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The default codec settings are used.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which does not specify any call or call transfer destination, since it only terminates calls. Its registration names are defined using a SEP7960BF0[00001-] sequence generating expression.
SK_SEP7960CF000001 uses the same settings as SK_SEP7960AF000001, except for the Dial Plan page, which specifies SK_SEP7960BF000001 as call destination. Its registration names are defined using a SEP7960CF0[00001-] sequence generating expression.
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Phone C of SK_SEP7960CF100001 dials the number of the call group compris-ing phones A and B and picks up the call.
Once connected, the two parties involved in the call, phones A and C, perform a voice session (Figure 4-64).
Figure 4-64. SK_036_7960_SM_US_5_Chs_IPv4_Static_Call_Group_Pickup Test Scenario
SK_SEP7960AF100001 configured settings are described in Table 4-71.
Phone A
Phone B
Phone C
Table 4-71. SK_SEP7960AF100001 Activity Test Settings
Category Settings
Scenario Editor The three channel test scenario is completely configured. On the Cisco CallManager, phones A and B are configured as part of one call group, while phone C is configured as part of another.
Phone A on the first scenario channel calls phone B using a Skinny Make Call script function, with phone B not answering the call.
Using a Skinny Call GPickUp procedure, phone C dials the group number of phones A and B and is able to pick up the call addressed to phone B.
Once connected, phones A and C perform media streaming.
Execution Settings The corresponding scenario channel is configured to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF1[00001-] sequence generating expression.
SK_SEP7960BF100001 is configured as a call destination.
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SK_037_7960_SM_US_5_Chs_IPv4_Static_Call_Pickup
This test, illustrating a call pickup procedure, comprises three VoIPSkinny Peer activities: phone A emulated by SK_SEP7960AF200001 establishes a call to phone B emulated by SK_SEP7960BF200001, which does not answer the call. Phone C of SK_SEP7960CF200001 presses the Call PickUp softkey and picks up the call.
Once connected, the two parties involved in the call, phones A and C, perform a voice session (Figure 4-65).
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The default codec settings are used.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF100001 uses the same settings as SK_SEP7960AF100001, except for the Dial Plan page, which does not specify any call or call transfer destination, since it only terminates calls. Its registration names are defined using a SEP7960BF1[00001-] sequence generating expression.
SK_SEP7960CF100001 uses the same settings as SK_SEP7960AF100001, except for the Dial Plan page, which does not specify a call destination. Its registration names are defined using a SEP7960CF1[00001-] sequence generating expression.
Table 4-71. SK_SEP7960AF100001 Activity Test Settings
Category Settings
Phone A
Phone B
Phone C
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Figure 4-65. SK_037_7960_SM_US_5_Chs_IPv4_Static_Call_Pickup Test Scenario
SK_SEP7960AF200001 configured settings are described in Table 4-72.
Table 4-72. SK_SEP7960AF200001 Activity Test Settings
Category Settings
Scenario Editor The three channel test scenario is completely configured. On the Cisco CallManager, phones A, B, and C are configured as part of the same call group.
Phone A on the first scenario channel calls phone B using a Skinny Make Call script function, with phone B not answering the call.
Phone C presses the Call PickUp softkey and picks up the call addressed to phone B.
Once connected, phones A and C perform media streaming.
Execution Settings The corresponding scenario channel is configured to use 1 alias/channel.
Dial Plan The registration names are defined using a SEP7960AF2[00001-] sequence generating expression.
SK_SEP7960BF200001 is configured as a call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
Codec Settings The default codec settings are used.
RTP Settings Since the activity performs RTP streaming, the Enable media on this activity option is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Note:
SK_SEP7960BF200001 uses the same settings as SK_SEP7960AF200001, except for the Dial Plan page, which does not specify any call or call transfer destination, since it only terminates calls. Its registration names are defined using a SEP7960BF2[00001-] sequence generating expression.
SK_SEP7960CF200001 uses the same settings as SK_SEP7960AF200001, except for the Dial Plan page, which does not specify any call or call transfer destination. Its registration names are defined using a SEP7960CF2[00001-] sequence generating expression.
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Mixed Skinny and SIP - SIP UAs
This group of test samples emulates Skinny and SIP phones that register with a Cisco CallManager as Skinny and SIP phones respectively and establish calls with each other. These samples are intended for running against a Cisco CallMa-nager 5.x or higher supporting the registration of Cisco SIP phones.
Test samples comprise both signaling-only and signaling with media calls.
MIX_023_7960_S0_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_10s
This test illustrates a mixed Skinny to SIP call procedure without media stream-ing. A number of 7960 Skinny phones emulated by the SK_SEP7960A0000001 activity register with the Cisco CallManager and then establish calls with the SIP phones emulated by the SIP_3000001 activity and registered with the Cisco Call-Manager as SIP phones. After the call is established the call, the connection is kept up for the duration of the Sleep functions, configured to 10 seconds.
The call procedure is executed once for the sustain time duration of the test.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-66.
Figure 4-66. MIX_023_7960_S0_US_3000_Chs_IPv4_Static_Sk_to_SIP_Call_10s Test Scenario
In order to be able to run the test on your machine, select the Skinny configura-tion settings described in Table 4-73 (SK_SEP7960A0000001) and the SIP set-tings given in Table 4-74 (SIP_3000001).
Note: For this tests category, all emulated SIP UAs are configured using consecutive IP addresses, the same port and consecutive phone number values.
Phone A
Phone B
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Table 4-73. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario is completely configured and comprises the phones registration followed by a call establishment procedure, a call sustain time without media streaming, and the call tear down.
On the call originating Skinny scenario channel, phone A registers with the Cisco CallManager as a Skinny phone and originates a call using the Skinny Make Call script function. The call duration is configured to 10 seconds using the Sleep function. The last scenario channel procedure is the Skinny End Call function that terminates the call.
On the call terminating SIP scenario channel, phone B registers with the Cisco CallManager as a SIP phone, then execute a SIP Receive Call procedure for answering the call. The Sleep function ensures a call duration of 10 seconds. The call is terminated using the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time and to use 1 alias/channel.
Dial Plan The Skinny registration names are defined using a SEP7960A00[00001-] sequence generating expression.
SIP_3000001:[5060-] is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-74. SIP_3000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario is completely configured and comprises the phones registration followed by a call establishment procedure, a call sustain time without media streaming, and the call tear down.
On the call originating Skinny scenario channel, phone A registers with the Cisco CallManager as a Skinny phone and originates a call using the Skinny Make Call script function. The call duration is configured to 10 seconds using the Sleep function. The last scenario channel procedure is the Skinny End Call function that terminates the call.
On the call terminating SIP scenario channel, phone B registers with the Cisco CallManager as a SIP phone, then execute a SIP Receive Call procedure for answering the call. The Sleep function ensures a call duration of 10 seconds. The call is terminated using the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use the consecutive IP addresses, the same port and consecutive phone numbers.
Dial Plan SIP endpoints phone numbers are defined using a 30[00001-] sequence generating expression.
Since this channel only terminates a call, no call destination or call transfer destination needs configured.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the SIP Settings area is specified using a [5060-] sequence generating expression.
In the Use Server area that needs to specify a SIP proxy server address and port, the address is that of the Cisco CallManager, and the port is 5060. Outbound proxy and registrar functionalities are also configured.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
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MIX_024_7960_SO_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_3min
This test is similar to the previous one, with the only difference that the call dura-tion is 3 minutes, configured using the Sleep script function.
MIX_025_7960_SO_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_30_min
This test is similar to the previous one, with the only difference that the call dura-tion is 30 minutes, configured using the Sleep script function.
MIX_026_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_SK_Call_10s
This test illustrates a mixed SIP to Skinny call procedure without media stream-ing. A number of 7960 SIP phones emulated by the SIP_3000001 activity regis-ter with the Cisco CallManager and then establish calls with the Skinny phones emulated by the SK_SEP7960A0000001 activity and registered with the Cisco CallManager as Skinny phones. After establishing the call, a call duration of 10 seconds is configured using Sleep script functions on both scenario channels.
The call procedure is executed once for the sustain time duration of the test.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-67.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and the following global scenario variables are initialized and can be used by all SIP channels:
• VoIP_Var0 is set to a 7960BBBB[0000-] sequence gen-erating expression. This variable is used by the SIP Make Registration procedure in the SIP Send Request (INVITE) script function for registering the SIP phone with a Cisco CallManager.
• VOIP_Var4 is set to ‘1’. This variable is used by the SIP Make Registration procedure to test whether the regis-tration is executed for every loop (for a value of ‘1’), or only for the first loop (for a value of ‘2’).
Table 4-74. SIP_3000001 Activity Test Settings
Category Settings
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Figure 4-67. MIX_026_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_Sk_Call_10s_ Test Scenario
In order to be able to run the test on your machine, select the Skinny configura-tion settings described in Table 4-75 (SK_SEP7960A0000001) and the SIP set-tings given in Table 4-76 (SIP_3000001).
Phone A
Phone B
Table 4-75. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario is completely configured and comprises the phones registration followed by a call establishment procedure, a call sustain time without media streaming, and the call tear down.
On the call originating scenario channel, SIP phone B registers with the Cisco CallManager as a SIP device and originates a calls towards the Skinny phone A. The established call is maintained active for a duration of 10 seconds using the Sleep script function. The last procedure is a call termination procedure for the receiving side.
On the call terminating scenario channel, Skinny phone A registers with the Cisco CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Skinny Wait Call and Skinny Answer Call script functions – for answering the call. The established call is maintained active for a duration of 10 seconds using the Sleep script function. The call is terminated by phone A using the Skinny End Call script function.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time and to use 1 alias/channel.
Dial Plan The Skinny registration names are defined using a SEP7960A00[00001-] sequence generating expression.
SInce this activity is terminating a call, no call destination needs configured.
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Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The CallManager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this activity performs no media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
Table 4-76. SIP_3000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure, a call sustain time without media streaming, and the call tear down.
On the call originating scenario channel, SIP phone B registers with the Cisco CallManager as a SIP device and originates a calls towards the Skinny phone A. The established call is maintained active for a duration of 10 seconds using the Sleep script function. The last procedure is a call termination procedure for the receiving side.
On the call terminating scenario channel, Skinny phone A registers with the Cisco CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Skinny Wait Call and Skinny Answer Call script functions – for answering the call. The established call is maintained active for a duration of 10 seconds using the Sleep script function. The call is terminated by phone A using the Skinny End Call script function.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use consecutive IP addresses, the same port and consecutive phone numbers.
Dial Plan SIP endpoints phone numbers are defined using a 30[00001-] sequence generating expression.
SK_SEP7960A0000001 is configured as call destination.
Table 4-75. SK_SEP7960A0000001 Activity Test Settings
Category Settings
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MIX_027_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_SK_Call_3min
This test is similar to the previous one, with the only difference that the call dura-tion is 3 minutes, configured using the Sleep script function.
MIX_028_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_SK_Call_30min
This test is similar to the previous one, with the only difference that the call dura-tion is 30 minutes, configured using the Sleep script function.
MIX_020_7960_SM_US_900_Chs_IPv4_Static_SIP_to_SK_Call_Voice_10s
This test illustrates a mixed SIP to Skinny call procedure with media streaming. A number of 7960 SIP phones emulated by the SIP_3000001 activity register with the Cisco CallManager and then establish calls with the Skinny phones emulated by the SK_SEP7960A0000001 activity and registered with the CallMa-nager. After establishing the call, the phones use the Voice Session script func-tion for bidirectional media streaming.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the IP Settings area is specified using a [5060-] sequence generating expression.
In the Use Server area that needs to specify a SIP proxy server address and port, the address is that of the Cisco CallManager, and the port is 5060. Outbound proxy and registrar functionalities are also configured.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this activity performs no media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and the following global scenario variables are initialized and can be read by all SIP channels:
• VoIP_Var0 is set to a 7960BBBB[0000-] sequence gen-erating expression. This variable is used by the SIP Make Registration procedure in the SIP Send Request (INVITE) script function for registering the SIP phone with a Cisco CallManager.
• VOIP_Var4 is set to ‘1’. This variable is used by the SIP Make Registration procedure to test whether the phones registration is executed for every loop (for a value of ‘1’), or only for the first loop (for a value of ‘2’).
Table 4-76. SIP_3000001 Activity Test Settings
Category Settings
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The call procedure is repeated once for the sustain time duration of the test.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-68.
Figure 4-68. MIX_020_7960_SM_US_900_Chs_IPv4_Static_SIP_to_Sk_Call_Voice_10s Test Scenario
In order to be able to run the test on your machine, select the Skinny configura-tion settings described in Table 4-77 (SK_SEP7960A0000001) and the SIP set-tings given in Table 4-78 (SIP_3000001).
Phone A
Phone B
Table 4-77. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario is completely configured and comprises the phones registration followed by a call establishment procedure, bidirectional media streaming and call tear down.
On the call originating scenario channel, SIP phone B registers with the CallManager as a SIP phone and originates a call. Media is exchanged using the Voice Session script function. The last procedure is a call termination procedure for the receiving side.
On the call terminating Skinny scenario channel, Skinny phone A registers with the CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. The Voice Session function is used to perform bidirectional media streaming. The call termination is started by phone A using the Skinny End Call script function.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time and to use 1 alias/channel.
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Dial Plan The Skinny registration names are defined using the SEP7960A00[00001-] sequence generating expression.
SInce this activity is terminating a call, no call destination and no transfer destination needs configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting is selected for Skinny traffic.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected.
The industry-standard Express Forwarding (0xA0) TOS/DSCP setting for RTP traffic is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
Table 4-78. SIP_3000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario is completely configured and comprises the phones registration followed by a call establishment procedure, bidirectional media streaming and call tear down.
On the call originating scenario channel, SIP phone B registers with the CallManager as a SIP phone and originates a call. Media is exchanged using the Voice Session script function. The last procedure is a call termination procedure for the receiving side.
On the call terminating Skinny scenario channel, Skinny phone A registers with the CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. The Voice Session function is used to perform bidirectional media streaming. The call termination is started by phone A using the Skinny End Call script function.
Execution Settings
The corresponding scenario channel is configured to execute once during the test sustain time and to use 1 alias/channel.
The emulated SIP phones use consecutive IP addresses, the same port and consecutive phone numbers.
Dial Plan The SIP endpoints phone numbers are defined using a 30[00001-] sequence generating expression.
SK_SEP7960A0000001 is configured as call destination.
Table 4-77. SK_SEP7960A0000001 Activity Test Settings
Category Settings
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MIX_021_7960_SM_US_900_Chs_IPv4_Static_SIP_to_SK_Call_Voice_3min
This test is similar to the previous one, with the only difference that the voice ses-sion has a duration of 3 minutes. The longer call duration is obtained by playing the selected wave file continuously for 3 minutes (Advanced Playback Settings tab of the Voice Session script function).
MIX_022_7960_SM_US_900_Chs_IPv4_Static_SIP_to_SK_Bulk_Call_Voice_30_min
This test is similar to the previous one, with the only difference that the voice ses-sion has a duration of 30 minutes. The longer call duration is obtained by playing the selected wave file continuously for 30 minutes (Advanced Playback Settings tab of the Voice Session script function).
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the SIP Settings area is specified using the [5060-] sequence generating expression.
In the Use Server area that needs to specify a SIP proxy server address and port, the address is that of the Cisco CallManager, and the port is 5060.
Outbound proxy and registrar functionalities are selected.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected.
Since in the Execution Settings page the SIP phones are configured to use consecutive IPs and the same port, the RTP Port field is specified using a single 10000 value.
Other Settings The IP version preference is set to IPv4, and the following global scenario variables are initialized and can be read by the SIP channels:
• VoIP_Var0 is set to the 7960BBBB[0000-] sequence gen-erating expression. This variable is used by the SIP Make Registration procedure in the SIP Send Request (INVITE) script function for registering the SIP phone with a Cisco CallManager.
• VOIP_Var4 is set to ‘1’. This variable is used by the SIP Make Registration procedure to test whether the current loop is the first loop or not. The phones registration can be executed for every loop (for a value of ‘1’), or only for the first loop (for a value of ‘2’).
Table 4-78. SIP_3000001 Activity Test Settings
Category Settings
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MIX_017_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_Call_Voice_10s
This test illustrates a mixed Skinny to SIP call procedure with media streaming. A number of 7960 Skinny phones emulated by the SK_SEP7960A0000001 activ-ity register with the Cisco CallManager and then establish calls with the SIP phones emulated by the SIP_3000001 activity and registered with the CallMan-ager as SIP phones. After establishing the call, the phones use the Voice Session script function for bidirectional media streaming.
The call procedure is repeated once for the sustain time duration of the test.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-69.
Figure 4-69. MIX_017_7960_SM_US_900_Chs_IPv4_Static_Sk_to_SIP_Call_Voice_10s Test Scenario
In order to be able to run the test on your machine, select the Skinny configura-tion settings described in Table 4-79 (SK_SEP7960A0000001) and the SIP set-tings given in Table 4-80 (SIP_3000001).
Phone A
Phone B
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Table 4-79. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure, bidirectional media streaming and call tear down.
On the call originating Skinny scenario channel, Skinny phone A registers with the CallManager and originates a call using the Skinny Make Call script function. Bidirectional media is exchanged using the Voice Session function. The last scenario channel procedure is the Skinny End Call function that terminates the call.
On the call terminating SIP scenario channel, SIP phone B registers with the CallManager as a SIP phone, then executes a SIP Receive Call procedure for answering the call. The Voice Session function performs bidirectional media streaming. The call is terminated using the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time and to use 1 alias/channel.
Dial Plan The Skinny registration names are defined using the SEP7960A00[00001-] sequence generating expression.
SIP_3000001:[5060] is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected.
The industry-standard Express Forwarding (0xA0) TOS/DSCP setting for RTP traffic is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-80. SIP_3000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure, bidirectional media streaming and call tear down.
On the call originating Skinny scenario channel, Skinny phone A registers with the CallManager and originates a call using the Skinny Make Call script function. Bidirectional media is exchanged using the Voice Session function. The last scenario channel procedure is the Skinny End Call function that terminates the call.
On the call terminating SIP scenario channel, SIP phone B registers with the CallManager as a SIP phone, then executes a SIP Receive Call procedure for answering the call. The Voice Session function performs bidirectional media streaming. The call is terminated using the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time and to use 1 alias/channel.
The emulated SIP phones use consecutive IP addresses, the same port and consecutive phone numbers.
Dial Plan The SIP endpoints phone numbers are defined using a 30[00001-] sequence generating expression.
Since this channel only terminates a call, no call destination needs configured.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the SIP Settings area is specified using a [5060-] sequence generating expression.
In the Use Server area that needs to specify a SIP proxy server address and port, the address is that of the Cisco CallManager, and the port is 5060. Outbound proxy and registrar functionalities are configured.
Codec Settings The default codec settings are used.
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MIX_018_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_Call_Voice_3min
This test is similar to the previous one, with the only difference that the voice ses-sion has a duration of 3 minutes. The longer call duration is obtained by playing the selected wave file continuously for 3 minutes (Advanced Playback Settings tab of the Voice Session script function).
MIX_019_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_Call_Voice_30_min
This test is similar to the previous one, with the only difference that the voice ses-sion has a duration of 30 minutes. The longer call duration is obtained by playing the selected wave file continuously for 30 minutes (Advanced Playback Settings tab of the Voice Session script function).
RTP Settings The Enable media on this activity option is selected.
Since in the Execution Settings page the SIP phones are configured to use consecutive IPs and the same port, the RTP Port field is specified using a single 10000 value.
Other Settings The IP version preference is set to IPv4, and the following global scenario variables are initialized and can be used by all SIP channels:
• VoIP_Var0 is set to the 7960BBBB[0000-] sequence generating expression. This variable is used by the SIP Make Registration procedure in the SIP Send Request (INVITE) script function for registering the SIP phone with a Cisco CallManager.
• VOIP_Var4 is set to ‘1’. This variable is used by the SIP Make Registration procedure to test whether the reg-istration is executed for every loop (for a value of ‘1’), or only for the first loop (for a value of ‘2’).
Table 4-80. SIP_3000001 Activity Test Settings
Category Settings
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MIX_031_7960_SO_US_75k_BHCA_IPv4_Static_Sk_to_SIP_Call
This test illustrates a signaling-only mixed Skinny to SIP call with a configured objective of 75000 calls/hour that is to be attained using 3000 channels. The Talk Time parameter is computed automatically based on the values of the BHCA value and the specified number of channels.
The 7960 Skinny phones emulated by the SK_SEP7960A0000001 activity regis-ter with the Cisco CallManager and then establish calls with the SIP phones emu-lated by the SIP_3000001 activity and registered with the CallManager as SIP phones. After the call is established, it is kept active for the duration of the com-puted Talk Time parameter by configuring the Sleep function using the $Talk-Time variable.
The call procedure is executed once for the entire test sustain time duration.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-70.
Figure 4-70. MIX_031_7960_S0_US_75k_BHCA_IPv4_Static_Sk_to_SIP_Call Test Scenario
The Skinny configuration settings are described in Table 4-81 (SK_SEP7960A0000001) and the SIP settings are given in Table 4-82 (SIP_3000001).
Phone B
Phone A
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Table 4-81. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure, a call sustain time without media streaming, and call tear down.
On the call originating Skinny scenario channel, Skinny phone A registers with the Cisco CallManager and originates a call using the Skinny Make Call script function. In the Sleep script function, the call duration is set to the value of the Talk Time parameter. The last scenario channel function is the Skinny End Call function that terminates the call.
On the call terminating SIP scenario channel, SIP phone B registers with the CallManager as a SIP phone, then executes a SIP Receive Call procedure for answering the call. The Sleep function is configured with a a duration equal to the Talk Time parameter. The call is terminated using the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time and to use 1 alias/channel.
Dial Plan The Skinny registration names are defined using the SEP7960A00[00001-] sequence generating expression.
SIP_3000001:[5060] is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-82. SIP_3000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure, a call sustain time without media streaming, and call tear down.
On the call originating Skinny scenario channel, Skinny phone A registers with the Cisco CallManager and originates a call using the Skinny Make Call script function. In the Sleep script function, the call duration is set to the value of the Talk Time parameter. The last scenario channel function is the Skinny End Call function that terminates the call.
On the call terminating SIP scenario channel, SIP phone B registers with the CallManager as a SIP phone, then executes a SIP Receive Call procedure for answering the call. The Sleep function is configured with a a duration equal to the Talk Time parameter. The call is terminated using the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use consecutive IP addresses, the same port and consecutive phone numbers.
Dial Plan SIP endpoints phone numbers are defined using a 30[00001-] sequence generating expression.
Since this channel only terminates a call, no call destination needs configured.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the SIP Settings area is specified using a [5060-] sequence generating expression.
In the Use Server area that needs to specify a SIP proxy server address and port, the address is that of the Cisco CallManager, and the port is 5060. Outbound proxy and registrar functionalities are also configured.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
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MIX_029_7960_SM_US_75k_BHCA_IPv4_Static_Sk_to_SIP_Call_Voice
The test is similar to the previous one, with the difference that the underlying test scenario uses Voice Session script functions instead of Sleep function. Both Voice Session functions are configured to perform media streaming for the dura-tion of the Talk Time parameter in the Listen and Advanced Playback Settings pages.
MIX_032_7960_SO_US_75k_BHCA_IPv4_Static_SIP_to_Sk_Call
This test illustrates a mixed SIP to Skinny call with a configured objective of 75000 calls/hour that are to be attained using 3000 channels. The Talk Time parameter is computed automatically based on the values of the BHCA value and the number of channels.
The 7960 Skinny phones emulated by the SK_SEP7960A0000001 activity regis-ter with the Cisco CallManager and receive the calls originated by the SIP phones emulated by the SIP_3000001 activity and registered with the Cisco Call-Manager as SIP phones. After the call is established, it is kept active for the dura-tion of the computed Talk Time parameter by configuring the Sleep function using the $TalkTime variable.
The call procedure is executed once for the sustain time duration.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-71.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and the following global scenario variables are initialized and can be used by all SIP channels:
• VoIP_Var0 is set to a 7960BBBB[0000-] sequence gen-erating expression. This variable is used by the SIP Make Registration procedure in the SIP Send Request (INVITE) script function for registering the SIP phone with a Cisco CallManager.
• VOIP_Var4 is set to ‘1’. This variable is used by the SIP Make Registration procedure to test whether the regis-tration is executed for every loop (for a value of ‘1’), or only for the first loop (for a value of ‘2’).
Note: Since media streaming is performed, both the Skinny and the SIP activity need to have the Enable media on this activity option selected. On the VoIPSIP activity, the RTP port is defined using a single 10000 value.
Table 4-82. SIP_3000001 Activity Test Settings
Category Settings
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Figure 4-71. MIX_032_7960_SO_US_75k_BHCA_IPv4_Static_SIP_to_Sk_Call Test Scenario
In order to be able to run the test on your machine, select the Skinny configura-tion settings described in Table 4-83 (SK_SEP7960A0000001) and the SIP set-tings given in Table 4-84 (SIP_3000001).
Phone A
Phone B
Table 4-83. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure, a call sustain time without media streaming, and call tear down.
On the call terminating Skinny scenario channel, Skinny phone A registers with the CallManager and receives the call using the Sk_Receive AnswerCall_HL procedure. In the Sleep script function, the call duration is set to the value of the Talk Time parameter. The last scenario channel function is the Skinny End Call function that terminates the call.
On the call originating SIP scenario channel, SIP phone B registers with the CallManager as a SIP phone, then executes a SIP Make Call procedure. The Sleep function is configured with a a duration equal to the Talk Time parameter. The call is terminated using the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
Dial Plan The Skinny registration names are defined using the SEP7960A00[00001-] sequence generating expression.
Since the channel only terminates a call, no call destination needs configured.
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Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
Table 4-84. SIP_3000001 Activity Test Settings
Category Settings
Scenario Editor The test scenario comprises the phones registration followed by a call establishment procedure, a call sustain time without media streaming, and call tear down.
On the call terminating Skinny scenario channel, Skinny phone A registers with the CallManager and receives the call using the Sk_Receive AnswerCall_HL procedure. In the Sleep script function, the call duration is set to the value of the Talk Time parameter. The last scenario channel function is the Skinny End Call function that terminates the call.
On the call originating SIP scenario channel, SIP phone B registers with the CallManager as a SIP phone, then executes a SIP Make Call procedure. The Sleep function is configured with a a duration equal to the Talk Time parameter. The call is terminated using the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time and to use 1 alias/channel.
The emulated SIP phones use consecutive IP address, the same port and consecutive phone numbers.
Dial Plan SIP endpoints phone numbers are defined using a 30[00001-] sequence generating expression.
SK_SEP7960A0000001 is configured as call destination.
Table 4-83. SK_SEP7960A0000001 Activity Test Settings
Category Settings
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MIX_030_7960_SM_US_75k_BHCA_IPv4_Static_SIP_to_Sk_Call_Voice
The test is similar to the previous one, with the difference that the underlying test scenario uses Voice Session script functions instead of Sleep function. Both Voice Session functions are configured to perform media streaming for the dura-tion of the Talk Time parameter in the Listen and Advanced Playback Settings pages.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the SIP Settings area is specified using a [5060-] sequence generating expression.
In the Use Server area that needs to specify a SIP proxy server address and port, the address is that of the CallManager configured for Skinny, and the port is 5060. Outbound proxy and registrar functionalities are also configured for the used SIP proxy server.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is not selected.
Other Settings The IP version preference is set to IPv4, and the following global scenario variables are initialized and can be used by all SIP channels:
• VoIP_Var0 is set to a 7960BBBB[0000-] sequence gen-erating expression. This variable is used by the SIP Make Registration procedure in the SIP Send Request (INVITE) script function for registering the SIP phone with a Cisco CallManager.
• VOIP_Var4 is set to ‘1’. This variable is used by the SIP Make Registration procedure to test whether the reg-istration is executed for every loop (for a value of ‘1’), or only for the first loop (for a value of ‘2’).
Note: Since media streaming is performed during the test, both the VoIPSkinny and VoIPSIP activity need to have the Enable media on this activity option selected. On the VoIPSIP activity, the RTP port is configured using a single 10000 value.
Table 4-84. SIP_3000001 Activity Test Settings
Category Settings
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Mixed Skinny and SIP - SIP Trunk
This group of sample configuration files are used for testing the SIP trunk sup-port implemented by the Cisco CallManager 4.x and newer.
Sample configurations, used for both signaling-only and signaling with media tests, comprise emulated Skinny IP phones registered with a Cisco CallManager that establish calls to SIP phones via a SIP trunk.
MIX_001_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_10s
This test illustrates a mixed Skinny to SIP call procedure without media stream-ing. The Skinny phones emulated by the SK_SEP7960A0000001 activity register with the Cisco CallManager and then establish calls with the SIP phones – emu-lated by the SIP_2000000001 activity – via a SIP trunk. After the call is estab-lished, the connection is kept up for the duration of the Sleep functions, configured to 10 seconds.
The call procedure runs once for the test duration.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-72.
Figure 4-72. MIX_001_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_10s Test Scenario
SK_SEP7960A0000001 and SIP_2000000001 configured settings are described in Table 4-85 and Table 4-86 respectively.
IMPORTANT: This test assumes that a SIP trunk has been configured between the Cisco CallManager and IxLoad, with the trunk configuration having the Significant Digits option configured to the All value.
Phone A
Phone B
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Table 4-85. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating Skinny scenario channel, phone A registers with the Cisco CallManager and originates a call to phone B via a SIP trunk configured on the CallManager. After call initiation using the Skinny Make Call script function, the call duration is configured to 10 seconds using the Sleep function. Finally phone A executes the Skinny End Call function that terminates the call.
On the call terminating SIP scenario channel, SIP phone B executes a SIP Receive Call procedure for answering the call. The Sleep function ensures a call duration of 10 seconds, eventually the SIP EndCall Receive procedure is used for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
Dial Plan The Skinny registration names are defined using a SEP7960A00[00001-] sequence generating expression.
SIP_2000000001:5060 is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-86. SIP_2000000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating Skinny scenario channel, phone A registers with the Cisco CallManager and originates a call to phone B via a SIP trunk configured on the CallManager. After call initiation using the Skinny Make Call script function, the call duration is configured to 10 seconds using the Sleep function.
Finally phone A executes the Skinny End Call function that terminates the call.
On the call terminating SIP scenario channel, SIP phone B executes a SIP Receive Call procedure for answering the call. The Sleep function ensures a call duration of 10 seconds, eventually the SIP EndCall Receive procedure is used for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use the same IP address and port, and consecutive phone numbers.
The Accept multiple channels sharing the same IP:port option is selected.
Dial Plan SIP phone numbers are defined using a 20000[00001-] sequence generating expression.
Since this channel only terminates a call, no call destination needs configured.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the IP Settings area is specified using a single 5060 value.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the SIP channels.
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MIX_002_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_3min
This test is similar to the previous one, with the only difference that the call dura-tion is 3 minutes, configured using the Sleep script function.
MIX_003_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_30min
This test is similar to the previous one, with the only difference that the call dura-tion is 30 minutes, configured using the Sleep script function.
MIX_004_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_10s
This test illustrates a mixed SIP to Skinny call procedure without media stream-ing. SIP phones emulated by the SIP_2000000001 activity establish calls via a SIP trunk to a Cisco CallManager, with whom the Skinny phones emulated by the SK_SEP7960A0000001 activity are registered. After call establishment, a call duration of 10 seconds is configured using Sleep script functions on both sce-nario channels.
The call procedure is repeated once for the test duration.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-73.
Figure 4-73. MIX_004_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_10s Test Scenario
SK_SEP7960A0000001 and SIP_2000000001 configured settings are described in Table 4-87 and Table 4-88 respectively.
IMPORTANT: This test assumes that a SIP trunk has been configured between the Cisco CallManager and IxLoad, with the trunk configuration having the Significant Digits option configured to the All value.
Phone A
Phone B
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Table 4-87. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, SIP phone B originates a call to Skinny phone A via a SIP trunk configured on the CallManager. After call initiation using the SIP MakeCall procedure, the call is maintained active for a duration of 10 seconds using the Sleep script function. The last SIP EndCall Receive procedure is a call termination procedure for the receiving side.
On the call terminating Skinny channel, Skinny phone A registers with the Cisco CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. The call is maintained active for a duration of 10 seconds using the Sleep script function, before it is terminated using the Skinny End Call script function.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
Dial Plan The Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
The Skinny registration names are defined using a SEP7960A00[00001-] sequence generating expression.
Since this activity only terminates a call, no call destination and no transfer destination needs configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The CallManager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this activity performs no media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-88. SIP_2000000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, SIP phone B originates a call to Skinny phone A via a SIP trunk configured on the CallManager. After call initiation using the SIP MakeCall procedure, the call is maintained active for a duration of 10 seconds using the Sleep script function. The last SIP EndCall Receive procedure is a call termination procedure for the receiving side.
On the call terminating Skinny channel, Skinny phone A registers with the Cisco CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. The call is maintained active for a duration of 10 seconds using the Sleep script function, before it is terminated using the Skinny End Call script function.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use the same IP address and same port, and consecutive phone numbers.
The Accept multiple channels sharing the same IP:port option is selected.
Dial Plan SIP phone numbers are defined using a 20000[00001-] sequence generating expression.
The configured call destination is the address of the CallManager and the 5080 port.
The override numbers of destination activity option is selected and the Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the IP Settings area is specified using a single 5060 value.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this activity performs no media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the SIP channels.
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MIX_005_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_3min
This test is similar to the previous one, with the only difference that the call dura-tion is 3 minutes, configured using the Sleep script function.
MIX_006_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_30min
This test is similar to the previous one, with the only difference that the call dura-tion is 30 minutes, configured using the Sleep script function.
MIX_012_7960_SM_US_900_Chs_IPv4_Static_SIP_to_SK_trunk_Call_Voice_10s
This test illustrates a mixed SIP to Skinny call scenario with media streaming. SIP phones emulated by the SIP_2000000001 activity establish calls via a SIP trunk to a Cisco CallManager, with whom the Skinny phones emulated by the SK_SEP7960A0000001 activity are registered. After establishing the call, the phones use the Voice Session script function for bidirectional media streaming.
The call scenario runs once for the duration of the test.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-74.
Figure 4-74. MIX_012_7960_SM_US_900_Chs_IPv4_Static_SIP_to_Sk_trunk_Call_Voice_10s Test Scenario
SK_SEP7960A0000001 and SIP_2000000001 configured settings are described in Table 4-89 and Table 4-90 respectively.
IMPORTANT: This test assumes that a SIP trunk has been configured between the Cisco CallManager and IxLoad, with the trunk configuration having the Significant Digits option configured to the All value.
Phone A
Phone B
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Table 4-89. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, SIP phone B originates a call to Skinny phone A via a SIP trunk configured on the CallManager. After call establishment using the SIP MakeCall procedure, media is exchanged using the Voice Session script function. The last procedure executed by SIP phone B is a call termination procedure for the receiving side.
On the call terminating scenario channel, Skinny phone A registers with the CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. The Voice Session function is used to perform bidirectional media streaming. Finally phone A executes the Skinny End Call script function for tearing down the call.
Execution Settings
The corresponding scenario channel is configured to execute once during the test sustain time.
Dial Plan The Skinny registration names are defined using the 7960AAAA[0000-] sequence generating expression.
The Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
Since this activity is terminating a call, no call destination needs configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting is selected for Skinny traffic.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected, with RTP port 10000 specified.
The industry-standard Express Forwarding (0xA0) TOS/DSCP setting for RTP traffic is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-90. VoIPSIPPeer1 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, SIP phone B originates a call to Skinny phone A via a SIP trunk configured on the CallManager. After call establishment using the SIP MakeCall procedure, media is exchanged using the Voice Session script function. The last procedure executed by SIP phone B is a call termination procedure for the receiving side.
On the call terminating scenario channel, Skinny phone A registers with the CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. The Voice Session function is used to perform bidirectional media streaming. Finally phone A executes the Skinny End Call script function for tearing down the call.
Execution Settings
The corresponding scenario channel is configured to execute once during the test duration.
The emulated SIP phones use the same IP address and port, and consecutive phone numbers.
The Accept multiple channels sharing the same IP:port option is selected.
Note: The Use consecutive value setting for the TCP/UDP port is intended for correct configuration of RTP ports.
Dial Plan SIP phone numbers are defined using a 20000[00001-] sequence generating expression.
The address and port 5080 of the Cisco CallManager are configured as call destination.
The override numbers of destination activity option is selected and the Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the IP Settings area is specified using a single 5060 value.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected.
The RTP Port field is specified using a [10000-2000,2] sequence generating expression.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the SIP channels.
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MIX_013_7960_SM_US_900_Chs_IPv4_Static_SIP_SK_trunk_Call_Voice_3min
This test is similar to the previous one, with the only difference that the voice ses-sion has a duration of 3 minutes. The longer call duration is obtained by playing the selected wave file continuously for 3 minutes (Advanced Playback Settings tab of the Voice Session script function).
MIX_014_7960_SM_US_900_Chs_IPv4_Static_SIP_SK_trunk_Call_Voice_30min
This test is similar to the previous one, with the only difference that the voice ses-sion has a duration of 30 minutes. The longer call duration is obtained by playing the selected wave file continuously for 30 minutes (Advanced Playback Settings tab of the Voice Session script function).
MIX_009_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunkCall_Voice_10s
This test illustrates a mixed Skinny to SIP call scenario with media streaming. A number of Skinny phones emulated by the SK_SEP7960A0000001 activity regis-ter with the Cisco CallManager and then originate calls to the SIP phones, emu-lated by the SIP_2000000001 activity, via a SIP trunk. After establishing the call, the phones use the Voice Session script function for bidirectional media stream-ing.
The call scenario runs once for the duration of the test.
The underlying two-channel test scenario involving Skinny phone A and SIP Trunk B is shown in Figure 4-75.
Figure 4-75. MIX_009_7960_SM_US_900_Chs_IPv4_Static_Sk_to_SIP_trunk_Call_Voice_10s Test Scenario
IMPORTANT: This test assumes that a SIP trunk has been configured between the Cisco CallManager and IxLoad, with the trunk configuration having the Significant Digits option configured to the All value.
Phone A
Phone B
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SK_SEP7960A0000001 and SIP_2000000001 configured settings are described in Table 4-91 and Table 4-92 respectively.
Table 4-91. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, Skinny phone A, registered with the CallManager, originates a call via a SIP trunk configured on the CallManager to SIP phone B. After call establishment using the Skinny Make Call script function, bidirectional media is exchanged. The last scenario channel procedure executed by phone A is the Skinny End Call function that terminates the call.
On the call terminating scenario channel, SIP phone B executes a SIP Receive Call procedure for answering the call. The Voice Session function performs bidirectional media streaming. Finally phone B executes the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding channel is configured to execute once during the test sustain time.
Dial Plan The Skinny registration names are defined using the SEP7960A00[00001-] sequence generating expression.
The Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
SIP_2000000001:5060 is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected and RTP port 10000 is specified.
The industry-standard Express Forwarding (0xA0) TOS/DSCP setting for RTP traffic is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-92. SIP_2000000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, Skinny phone A, registered with the CallManager, originates a call via a SIP trunk configured on the CallManager to SIP phone B. After call establishment using the Skinny Make Call script function, bidirectional media is exchanged. The last scenario channel procedure executed by phone A is the Skinny End Call function that terminates the call.
On the call terminating scenario channel, SIP phone B executes a SIP Receive Call procedure for answering the call. The Voice Session function performs bidirectional media streaming. Finally phone B executes the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use the same IP address and port, and consecutive phone numbers.
The Accept multiple channels sharing the same IP:port option is selected.
Note: The Use consecutive value setting for the TCP/UDP port is intended for correct configuration of RTP ports.
Dial Plan SIP phone numbers are defined using a 20000[00001-] sequence generating expression.
Since this activity only terminates a call, no call destination needs configured.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the IP Settings area is specified using a single 5060 value.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected.
The RTP Port field needs specified using the [10000-20000, 2] sequence generating expression.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the SIP channels.
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MIX_010_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunk_Call_Voice_3min
This test is similar to the previous one, with the only difference that the voice ses-sion has a duration of 3 minutes. The longer call duration is obtained by playing the selected wave file continuously for 3 minutes (Advanced Playback Settings tab of the Voice Session script function).
MIX_011_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunk_Call_Voice_30min
This test is similar to the previous one, with the only difference that the voice ses-sion has a duration of 30 minutes. The longer call duration is obtained by playing the selected wave file continuously for 30 minutes (Advanced Playback Settings tab of the Voice Session script function).
MIX_008_7960_SO_US_80k_BHCA_IPv4_Static_SIP_to_SK_trunk_Bulk_Call
This test illustrates a mixed SIP to Skinny call procedure without media stream-ing, having a configured objective of 80000 calls/hour that is to be attained using 5000 channels. The Talk Time parameter is computed automatically based on the values of the BHCA value and the specified number of channels.
The SIP phones emulated by the SIP_2000000001 activity establish calls via a SIP trunk to a Cisco CallManager, with whom the Skinny phones emulated by the SK_SEP7960A0000001 activity are registered. After the call is established, it is kept active for the duration of the computed Talk Time parameter by configur-ing the Sleep function using the $TalkTime variable.
The test is run once for the sustain time duration.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-76.
IMPORTANT: This test assumes that a SIP trunk has been configured between the Cisco CallManager and IxLoad, with the trunk configuration having the Significant Digits option configured to the All value.
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Figure 4-76. MIX_008_7960_SO_US_80k_BHCA_IPv4_Static_SIP_to_SK_trunk_Bulk_Call Test Scenario
SK_SEP7960A0000001 and SIP_2000000001 configured settings are described in Table 4-93 and Table 4-94 respectively.
Phone A
Phone B
Table 4-93. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, SIP phone B originates a call to Skinny phone A via a SIP trunk configured on the CallManager. After call initiation using the SIP MakeCall procedure, the Sleep script function configures the call duration to the value of the Talk Time parameter. The last procedure executed by SIP phone B is a call termination procedure for the receiving side.
On the call terminating scenario channel, Skinny phone A registers with the CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. In the Sleep script function, the call duration is set to the value of the Talk Time parameter. Finally phone A executes the Skinny End Call script function for tearing down the call.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
Dial Plan The Skinny registration names are defined using the SEP7960A00[00001-] sequence generating expression.
The Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
Since the channel only terminates a call, no call destination needs configured.
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Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
Table 4-94. SIP_2000000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, SIP phone B originates a call to Skinny phone A via a SIP trunk configured on the CallManager. After call initiation using the SIP MakeCall procedure, the Sleep script function configures the call duration to the value of the Talk Time parameter. The last procedure executed by SIP phone B is a call termination procedure for the receiving side.
On the call terminating scenario channel, Skinny phone A registers with the CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. In the Sleep script function, the call duration is set to the value of the Talk Time parameter. Finally phone A executes the Skinny End Call script function for tearing down the call.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use the same IP address and port, and consecutive phone numbers.
The Accept multiple channels sharing the same IP:port option is selected.
Dial Plan SIP phone numbers are defined using a 20000[00001-] sequence generating expression.
The address of the Cisco CallManager and port 5080 are configured as call destination.
The override numbers of destination activity option is selected and the Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
Table 4-93. SK_SEP7960A0000001 Activity Test Settings
Category Settings
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MIX_016_7960_SM_US_75k_BHCA_IPv4_Static_SIP_to_SK_trunk_Call_Voice
This test illustrates a mixed SIP to Skinny call procedure with media streaming. SIP phones emulated by the SIP_2000000001 activity establish calls via a SIP trunk to a Cisco CallManager, with whom the Skinny phones emulated by the SK_SEP7960A0000001 activity are registered. After establishing the call, the phones use the Voice Session script function for bidirectional media streaming. Both Voice Session functions are configured to perform media streaming for the duration of the Talk Time parameter in the Listen and Advanced Playback Set-tings function pages.
The call procedure is repeated once for the test duration.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-75.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the SIP Settings area is specified using a single 5060 value.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the SIP channels.
IMPORTANT: This test assumes that a SIP trunk has been configured between the Cisco CallManager and IxLoad, with the trunk configuration having the Significant Digits option configured to the All value.
Table 4-94. SIP_2000000001 Activity Test Settings
Category Settings
Phone A
Phone B
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Figure 4-77. MIX_016_7960_SM_US_75k_BHCA_IPv4_Static_SIP_to_Sk_trunk_Call_Voice Test Scenario
SK_SEP7960A0000001 and SIP_2000000001 configured settings are described in Table 4-91 and Table 4-92 respectively.
Table 4-95. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, SIP phone B originates a call to Skinny phone A via a SIP trunk configured on the CallManager. After call establishment using the SIP MakeCall procedure, media is exchanged using the Voice Session script function. The last procedure executed by SIP phone B is a call termination procedure for the receiving side.
On the call terminating scenario channel, Skinny phone A registers with the CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. The Voice Session function is used to perform bidirectional media streaming. Finally phone A executes the Skinny End Call script function for tearing down the call.
Execution Settings The corresponding channel is configured to execute once during the test sustain time.
Dial Plan The Skinny registration names are defined using the SEP7960A00[00001-] sequence generating expression.
The Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
No call destination is configured.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected and RTP port 10000 is specified.
The industry-standard Express Forwarding (0xA0) TOS/DSCP setting for RTP traffic is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-96. SIP_2000000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, SIP phone B originates a call to Skinny phone A via a SIP trunk configured on the CallManager. After call establishment using the SIP MakeCall procedure, media is exchanged using the Voice Session script function. The last procedure executed by SIP phone B is a call termination procedure for the receiving side.
On the call terminating scenario channel, Skinny phone A registers with the CallManager and executes a Sk_Receive AnswerCall_HL procedure – a wrapping of the common Wait Call and Answer Call script functions – for answering the call. The Voice Session function is used to perform bidirectional media streaming. Finally phone A executes the Skinny End Call script function for tearing down the call.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use the same IP address and port, and consecutive phone numbers.
The Accept multiple channels sharing the same IP:port option is selected.
Note: The Use consecutive value setting for the TCP/UDP port is intended for correct configuration of RTP ports.
Dial Plan SIP phone numbers are defined using a 20000[00001-] sequence generating expression.
The address of the Cisco CallManager and port 5080 are configured as call destination.
The override numbers of destination activity option is selected and the Skinny phone numbers are specified using a 16[00001-] sequence generation expression.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the SIP Settings area is specified using a single 5060 value.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected.
The RTP Port field is specified using a [10000-2000,2] sequence generating expression.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the SIP channels.
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MIX_007_7960_SO_US_80k_BHCA_IPv4_Static_SK_to_SIP_trunk_Bulk_Call
This test illustrates a mixed Skinny to SIP call procedure without media stream-ing, having a configured objective of 80000 calls/hour that is to be attained using 3000 channels. The Talk Time parameter is computed automatically based on the values of the BHCA value and the number of channels.
The Skinny phones emulated by the SK_SEP7960A0000001 activity register with a Cisco CallManager and then originate calls to the SIP phones, emulated by the SIP_2000000001 activity, via a SIP trunk. After the call is established, it is kept active for the duration of the computed Talk Time parameter by configuring the Sleep function using the $TalkTime variable.
The call procedure is run once for the test duration.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-78.
Figure 4-78. MIX_007_7960_SO_US_80k_BHCA_IPv4_Static_SK_to_SIP_trunk_Bulk_Call Test Scenario
SK_SEP7960A0000001 and SIP_2000000001 configured settings are described in Table 4-97 and Table 4-98 respectively.
IMPORTANT: This test assumes that a SIP trunk has been configured between the Cisco CallManager and IxLoad, with the trunk configuration having the Significant Digits option configured to the All value.
Phone A
Phone B
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Table 4-97. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, Skinny phone A registers with the CallManager and originates a call via the SIP trunk configured on the CallManager to phone B using the Skinny Make Call script function. After call initiation using the Skinny Make Call function, the Sleep function configures the call duration to the value of the Talk Time parameter. The last scenario channel procedure executed by phone A is the Skinny End Call function that terminates the call.
On the call terminating scenario channel, phone B executes a SIP Receive Call procedure for answering the call. In the Sleep script function, the call duration is set to the value of the Talk Time parameter. Finally phone B executes the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
Dial Plan The Skinny registration names are defined using the SEP7960A00[00001-] sequence generating expression.
SIP_2000000001:5060 is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting for Skinny traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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Table 4-98. SIP_2000000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, Skinny phone A registers with the CallManager and originates a call via the SIP trunk configured on the CallManager to phone B using the Skinny Make Call script function. After call initiation using the Skinny Make Call function, the Sleep function configures the call duration to the value of the Talk Time parameter. The last scenario channel procedure executed by phone A is the Skinny End Call function that terminates the call.
On the call terminating scenario channel, phone B executes a SIP Receive Call procedure for answering the call. In the Sleep script function, the call duration is set to the value of the Talk Time parameter. Finally phone B executes the SIP EndCall Receive procedure for the call terminating side.
Execution Settings The corresponding scenario channel is configured to execute once during the test sustain time.
The emulated SIP phones use the same IP address and port, and consecutive phone numbers.
The Accept multiple channels sharing the same IP:port option is selected.
Dial Plan SIP phone numbers are defined using a 20000[00001-] sequence generating expression.
Since this channel only terminates a call, no call destination needs configured.
SIP Settings The Enable signaling on this activity option is selected for the SIP functions to be executed.
The Port field in the SIP Settings area is specified using a single 5060 value.
The industry-standard Class 3 (0x60) TOS/DSCP setting for SIP traffic is selected.
Codec Settings The default codec settings are used.
RTP Settings Since this test does not perform media streaming, the Enable media on this activity option is de-selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the SIP channels.
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MIX_015_7960_SM_US_75k_BHCA_IPv4_Static_Sk_to_SIP_trunk_Call_Voice
This test illustrates a mixed Skinny to call SIP procedure with media streaming.
The Skinny phones emulated by the SK_SEP7960A0000001 activity register with a Cisco CallManager and then originate calls to the SIP phones, emulated by the SIP_2000000001 activity, via a SIP trunk. After establishing the call, the phones use the Voice Session script function for bidirectional media streaming. Both Voice Session functions are configured to perform media streaming for the dura-tion of the Talk Time parameter in the Listen and Advanced Playback Settings script function pages.
The call procedure is repeated once for the test duration.
The underlying two-channel test scenario involving Skinny phone A and SIP phone B is shown in Figure 4-74.
Figure 4-79. MIX_015_7960_SM_US_75k_BHCA_IPv4_Static_Sk_to_SIP_trunk_Call_Voice Test Scenario
SK_SEP7960A0000001 and SIP_2000000001 configured settings are described in Table 4-89 and Table 4-90 respectively.
IMPORTANT: This test assumes that a SIP trunk has been configured between the Cisco CallManager and IxLoad, with the trunk configuration having the Significant Digits option configured to the All value.
Phone A
Phone B
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Table 4-99. SK_SEP7960A0000001 Activity Test Settings
Category Settings
Scenario Editor On the call originating scenario channel, Skinny phone A registers with the CallManager and originates a call to SIP phone B via a SIP trunk configured on the CallManager.
After call initiation using the Skinny Make Call function, the Voice Session function is used to perform bidirectional media streaming for the duration of the Talk Time parameter.
The last scenario channel procedure executed by phone A is the Skinny End Call function that terminates the call.
On the call terminating scenario channel, phone B executes a SIP Receive Call procedure for answering the call. After call establishment, media is exchanged using the Voice Session script function. Finally phone B executes the SIP EndCall Receive procedure for the call terminating side.
Execution Settings
The corresponding scenario channel is configured to execute once during the test sustain time.
Dial Plan The Skinny registration names are defined using the 7960AAAA[0000-] sequence generating expression.
SIP_2000000001:5060 is configured as call destination.
Skinny Settings The Enable signaling on this activity option is selected for the Skinny functions to be executed.
The Call Manager IP address and port need to be configured in the Call Managers area.
The industry-standard Class 3 (0x60) TOS/DSCP setting is selected for Skinny traffic.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected and RTP port 10000 is specified.
The industry-standard Express Forwarding (0xA0) TOS/DSCP setting for RTP traffic is selected.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized for the Skinny channels.
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H.323 Sample Test Configurations and Test Scenarios
Using the functions from the VoIP H.323 test library, you can generate and exe-cute a large number of originator/answerer scenario configurations that comply with the H.323 protocol. You can use one of the predefined test scenarios described in this chapter or create a new one, map it to VoIP H.323 activities, and start the execution.
This section describes the pre-defined IxLoad Voice Plug-in H.323 available sample test configurations (RXFs) and their associated test scenarios.
VoIP H.323 Test Configurations
The following sample H.323 test configuration files are contained in the IxLoad installation kit:
Note: For a complete description of the supported H.323 test library functions, refer to the VoIP H323 Functions Library on page 3-118.
Note: Sample tests follow a naming convention that comprises the test type (VH for VoIP H323), an index, a test configuration (B2B for Back-to-Back or GK for running against a Gatekeeper), a protocol version (IPv4 or IPv6), a test features description ( NC for a normal call, FC for a call using a FastConnect procedure, T for a call using a tunneling procedure, and PC for a call using a parallel procedure) and a short scenario description, such as for example VH_001_B2B_H323v4_NC_Basic_Call.
Note: Most of the provided sample test configurations share the following settings:
• The used H.225 and H245 protocol versions are 0.0.8.2250.0.5 and 0.0.8.245.0.9 respectively
• The Graceful ramp down option is selected
• The same terminal type (Terminal entity without MC) and bandwidth (64kbps) settings are used
• The used Q.931 settings are 1[00000-] and Caller[00000-] for the call origi-nator and 2[00000-] and Called[00000-] for the call terminator activities.
• The Send Alert and Send Call Processing options are selected for call ter-minating activities
• The default codecs and terminal capabilities settings comprise the G.711 ulaw and G.711 alaw codecs
• The RTP port is defined based on a sequence generator expression for all H323 activities linked to scenario channels that use RTP functions.
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VH_001_B2B_H323v4_NC_Basic_Call
This test which runs in B2B mode comprises two VoIP H323 activities, Make_Call and Receive_Call, that simulate a number of N phones on each side (N = 100, the test objective value) performing an H323 call without media exchange.
Make_Call is linked to a test scenario channel that originates the call, remains idle for a duration of time specified by the Sleep script function, and then discon-nects, as shown in Figure 4-80. Receive_Call executes the corresponding call receiving functions flow.
Figure 4-80. VH_001_B2B_H323v4_NC_Basic_Call Test Scenario
Make_Call configured settings are described in Table 4-100:
Terminal A
Terminal B
Table 4-100. Make_Call Activity Test Settings
Category Settings
Scenario On the call originating scenario channel, H323 terminal A originates a call to H323 terminal B. After call establishment, the call is maintained active for a duration configured by the Sleep function. Eventually it is terminated using an H323 End Call function.
On the call terminating scenario channel, terminal B executes an H323 Receive Call procedure for answering the call and finally the H323EndCall procedure for the call terminating side.
Execution The corresponding scenario channel is configured to execute repeatedly during the test sustain time.
The simulated H323 terminals use consecutive phone numbers (per activity).
Dial Plan The originating H323 phone numbers are defined using a 160[00000000-] sequence generator expression. The Receive_Call activity is configured as call destination.
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VH_002_B2B_H323v4_NC_Basic_Call_with_RTP
This test which runs in B2B mode comprises two VoIPH323 activities, Make_Call and Receive_Call, that perform an H323 call with bidirectional media exchange.
Make_Call is linked to a test scenario channel that originates the call, performs bidirectional media exchange, and then disconnects, as shown in Figure 4-81. Receive_Call executes the call originating functions flow.
Figure 4-81. VH_001_B2B_H323v4_NC_Basic_Call Test Scenario
Make_Call configured settings are described in Table 4-101:
H323 The Enable signaling on this activity option is selected for the H323 functions to be executed.
Terminal Capability
The default capabilities settings are used.
Codecs The default codec settings are used.
RTP Since this channel does not use any RTP functions, the Enable media on this activity option is not selected.
Other The IP version preference is set to IPv4, and no scenario variables need to be initialized for the H323 channels.
Note: The Receive_Call activity is configured in a similar way, except that it does not need to specify a call destination.
Table 4-100. Make_Call Activity Test Settings
Category Settings
Terminal A
Terminal B
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VH_003_B2B_H323v4_FC_Basic_Call_with_RTP
This test is similar to the previous VH_002_B2B_H323v4_NC_Basic_Call_with_RTP test, with the difference that the call is established using the the FastStart and Tunneling procedures config-ured in the H323 page.
VH_004_B2B_H323v4_PC_Basic_Call_with_RTP
This test is similar to the previous VH_002_B2B_H323v4_NC_Basic_Call_with_RTP test, with the difference that the call uses the FastStart, Tunneling, and Parallel procedures configured in the H323 page.
Table 4-101. Make_Call Activity Test Settings
Category Settings
Scenario On the call originating scenario channel, H323 terminal A originates a call to H323 terminal B. After call establishment, media is exchanged bidirectionally using the VoiceSession script function.
On the call terminating scenario channel, terminal B executes a H323 Receive Call procedure for answering the call.
Execution The corresponding scenario channel is configured to execute repeatedly for the test sustain time.
The emulated H323 terminals use consecutive phone numbers (per activity).
Dial Plan The originating H323 phone numbers are defined using a 160[00000000-] sequence generator expression. The Receive_Call activity is configured as call destination.
H323 The Enable signaling on this activity option is selected for the H323 functions to be executed.
Terminal Capability
The default capabilities settings are used.
Codecs The default codec settings are used.
RTP The Enable media on this activity option is selected.
Other The IP version preference is set to IPv4, and no scenario variables need to be initialized for the H323 channels.
Note: The Receive_Call activity is configured in a similar way, except that it does not need to specify a call destination.
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VH_005_B2B_H323v4_T_Basic_Call_with_RTP
This test is similar to the previous VH_002_B2B_H323v4_NC_Basic_Call_with_RTP test, with the difference that the call uses the Tunneling procedure configured in the H323 page.
VH_006_B2B_H323v4_NC_Make_Call_with_RTP
This test comprises a single VoIPH323 activity, Make_Call, that simulates a number of H323 phones (N = 100, the test objective value) originating an H323 call to another H323 device. After the call is established, media is exchanged bidirectionally.
Make_Call configured settings are described in Table 4-101:
VH_007_B2B_H323v4_FC_Make_Call_with_RTP
This test is similar to the previous VH_006_B2B_H323v4_NC_Make_Call_with_RTP test, with the difference that the call uses the FastStart and Tunneling procedures configured in the H323 page.
Table 4-102. Make_Call Activity Test Settings
Category Settings
Scenario H323 terminal A originates a call to another H323 terminal. After call establishment, media is exchanged using the Voice Session script function.
Execution The corresponding scenario channel is configured to execute continuously during the test sustain time.
The emulated H323 terminals use consecutive phone numbers (per activity).
Dial Plan The originating H323 phone numbers are defined using a 160[00000000-] sequence generator expression.
The call destination is specified using an IP address.
H323 The Enable signaling on this activity option is selected for the H323 functions to be executed.
Terminal Capability
The default capabilities settings are used.
Codecs The default codec settings are used.
RTP The Enable media on this activity option is selected.
Other The IP version preference is set to IPv4, and no scenario variables need to be initialized for the simulated H323 channel.
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VH_008_B2B_H323v4_PC_Make_Call_with_RTP
This test is similar to the previous VH_006_B2B_H323v4_NC_Make_Call_with_RTP test, with the difference that the call uses the Enable FastStart, Enable Tunneling, and Enable Parallel H245 options configured in the H323 page.
VH_009_B2B_H323v4_NC_Receive_Call_with_RTP
This test comprises a single VoIPH323 activity, Receive_Call, that terminates an H323 call originating from a H323 device. After the call is established, media is exchanged bidirectionally between the two terminals.
Receive_Call configured settings are described in Table 4-103:
VH_010_B2B_H323v4_FC_Receive_Call_with_RTP
This test is similar to the previous VH_009_B2B_H323v4_NC_Receive_Call_with_RTP test, with the difference that the call uses the FastStart and Tunneling procedures configured in the H323 page.
Table 4-103. Make_Call Activity Test Settings
Category Settings
Scenario The simulated H323 terminal terminates an incoming call from another H323 terminal using a H323 Receive Call function. After call establishment, media is exchanged using the Voice Session script function. Eventually it waits for the other party to disconnect and tears down the call.
Execution The corresponding scenario channel is configured to execute continuously during the test sustain time.
The emulated H323 terminals use consecutive phone numbers.
Dial Plan The H323 phone numbers are defined using a 160[00000000-] sequence generator expression. No call destination is specified.
H323 The Enable signaling on this activity option is selected for the H323 functions to be executed. The Send Call Alerting and Send Call Proceeding options are selected.
Terminal Capability
The default capabilities settings are used.
Codecs The default codec settings are used.
RTP The Enable media on this activity option is selected.
Other The IP version preference is set to IPv4, and no scenario variables need to be initialized for the simulated H323 channel.
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VH_011_B2B_H323v4_PC_Receive_Call_with_RTP
This test is similar to the previous VH_009_B2B_H323v4_NC_Receive_Call_with_RTP test, with the difference that the call uses the Enable FastStart, Enable Tunneling, and Enable Parallel H245 options configured in the H323 page.
VH_012_B2B_H323v4_NC_Basic_Call_with_RTP_Trunk
This test is similar to the previous VH_006_B2B_H323v4_NC_Make_Call_with_RTP test, with the following dif-ference at activity configuration level:
• All the simulated originating H323 endpoints (Make_Call activity) use a sin-gle IP address.
• The call destination is configured to a single IP address specified in the Dial Plan page of the Make_Call activity.
• All terminating H323 endpoints (Receive_Call activity) use another unique IP address, the same at that configured as call destination by the call-originating H323 endpoints.
VH_013_B2B_H323v4_FC_Basic_Call_with_RTP_Trunk
This test is similar to the previous VH_012_B2B_H323v4_NC_Basic_Call_with_RTP_Trunk test, with the differ-ence that the call uses the FastConnect procedure configured in the H323 page.
VH_014_B2B_H323v4_NC_Basic_Call_with_RTP_all_codecs
This test is similar to the previous VH_006_B2B_H323v4_NC_Make_Call_with_RTP test, with the difference that all supported codecs are configured in the H323 page and a custom capability descriptor is selected in the Terminal Capabilities page.
VH_015_B2B_H323v4_NC_Basic_Call_with_RTP_QoV
This test is similar to the previous VH_006_B2B_H323v4_NC_Make_Call_with_RTP test, with the difference that the test computes PESQ and P56 QoV scores for a number of 100 channels. The test has the Enable Qov option configured in the RTP page and the FemaleMale_Mix1 clip on the VoiceSession script functions.
VH_016_B2B_H323v4_NC_Basic_Call_with_HwRTP
This test is similar to the previous VH_006_B2B_H323v4_NC_Make_Call_with_RTP test, with the difference that
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the Enable Hw Acceleration option is selected in the H323 page of both origi-nating and terminating activities.
VH_017_B2B_H323v4_NC_Basic_Call_with_HwRTP_10GA_Trunk
This test is similar to the previous VH_006_B2B_H323v4_NC_Make_Call_with_RTP test, with the difference that the Allow multiple aggregated 1G ports option is selected in the Test Options configuration page.
VH_018_GK_H323v4_NC_Basic_Call
This test, which runs against a GK, comprises two VoIP H323 activities, Make_Call and Receive_Call, that simulate a number of N phones (N = 100, the test objective value) performing an H323 call without media exchange.
Make_Call is linked to a test scenario channel that originates the call, remains idle for a user-configured duration of time, and then disconnects, as shown in Figure 4-82. Receive_Call executes the call originating functions flow.
Figure 4-82. VH_001_B2B_H323v4_NC_Basic_Call Test Scenario
Make_Call configured settings are described in Table 4-104:
Terminal A
Terminal B
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VH_019_GK_H323v4_NC_Basic_Call_with_RTP
This test is similar to the previous VH_018_GK_H323v4_NC_Basic_Call test, with the difference that after call establishment, bidirectional media transfer is performed using the VoiceSession script function. As such, the Enable media on this activity option is selected for both the Make_Call and the Receive_Call activities.
Table 4-104. Make_Call Activity Test Settings
Category Settings
Scenario On the call originating scenario channel, H323 terminal A registers with a GK and originates a call to H323 terminal B. After call establishment, the terminal stays idle for a configured period of time, and then disconnects the call. Eventually the terminal unregisters with the GK.
On the call terminating scenario channel, terminal B first registers with the GK and then executes a H323 Receive Call procedure for answering the call.
Execution The corresponding scenario channel is configured to execute repeatedly during the test sustain time.
The simulated H323 terminals use consecutive phone numbers (per activity).
Dial Plan The originating H323 phone numbers are defined using a 160[00000000-] sequence generator expression. The Receive_Call activity is configured as call destination.
H323 The Enable signaling on this activity option is selected for the H323 functions to be executed.
The Enable RAS, Use Registration parameters, Use Gatekeeper for admission, Enable Disengage, Enable Keep-Alive registration options are selected.
The IP address of the GK the test is running against needs specified in the Gatekeeper area.
Terminal Capability
The default capabilities settings are used.
Codecs The default codec settings are used.
RTP The Enable media on this activity option is not selected.
Other The IP version preference is set to IPv4, and no scenario variables need to be initialized for the H323 channels.
Note: The Receive_Call activity is configured in a similar way, except that it does not need to specify a call destination.
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VH_020_GK_H323v4_FC_Basic_Call_with_RTP
This test is similar to the previous VH_019_GK_H323v4_NC_Basic_Call_with_RTP test, with the difference that the call uses the Enable FastStart and Enable Tunneling options configured in the H323 page. After call establishment, bidirectional media transfer is per-formed using the VoiceSession script function.
VH_021_GK_H323v4_PC_Basic_Call_with_RTP
This test is similar to the previous VH_019_GK_H323v4_NC_Basic_Call_with_RTP test, with the difference that the call uses the Enable FastStart, Enable Tunneling, and Enable Parallel H245 options configured in the H323 page. After call establishment, bidirec-tional media transfer is performed using the VoiceSession script function.
VH_022_GK_H323v4_T_Basic_Call_with_RTP
This test is similar to the previous VH_019_GK_H323v4_NC_Basic_Call_with_RTP test, with the difference that the call uses the Enable Tunneling option configured in the H323 page. After call establishment, bidirectional media transfer is performed using the VoiceSes-sion script function.
VH_023_GK_H323v4_NC_Basic_Call_with_RTP_Trunk
This test comprises two VoIP H323 activities, Make_Call and Receive_Call, that simulate a number of N (N = 100, the test objective value) H323 terminals per-forming an H323 call with media exchange. The test runs against a user-specified Gatekeeper as the DUT.
The Make_Call configured settings are described in Table 4-105:
Table 4-105. Make_Call Activity Test Settings
Category Settings
Scenario H323 terminal A registers with a GK and then originates a call to the H323 terminal B. After call establishment, media is exchanged bidirectionally using the Voice Session script function. Eventually terminal A tears down the call.
Execution The corresponding scenario channel is configured to execute continuously during the test sustain time.
The emulated H323 terminals use consecutive phone numbers (per activity).
Dial Plan The H323 phone numbers are defined using a 160[00000000-] sequence generator expression. Receive_Call is configured as call destination.
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VH_024_GK_H323v4_FC_Basic_Call_with_RTP_Trunk
This test is similar to the previous VH_019_GK_H323v4_NC_Basic_Call_with_RTP test, with the difference that the call uses the FastStart and Enable Tunneling options configured in the H323 page.
VH_025_GK_H323v4_NC_Basic_Call_with_RTP_all_codecs
This test is similar to the previous VH_019_GK_H323v4_NC_Basic_Call_with_RTP test, with the difference that all supported codecs are configured in the H323 page and a custom capability descriptor is selected in the Terminal Capabilities page.
VH_026_GK_H323v4_NC_Basic_Call_with_RTP_QoV
This test is similar to the previous VH_019_GK_H323v4_NC_Basic_Call_with_RTP test, with the difference that the test computes PESQ and P56 QoV metrics. The test has the Enable QoV option configured in the RTP page of both originating and terminating activities, and the FemaleMale_Mix1 clip on the VoiceSession script functions.
H323 The Enable signaling on this activity option is selected for the H323 functions to be executed. The GK IP address is configured in the Gatekeeper area.
Terminal Capability
The default capabilities settings are used.
Codecs The default codec settings are used.
RTP The Enable media on this activity option is selected.
Since all RTP endpoints use a single IP address, a [10000-65535, 2] sequence generating expression needs specified in the RTP Port field.
Other The IP version preference is set to IPv4, and no scenario variables need to be initialized for the H323 channels.
Note: The Receive_Call activity is configured in a similar way, except that it does not need to specify a call destination.
Note: This test is different from other sample tests in that the Make_Call simulated H323 endpoints use a single IP address. This also holds true for the Make_Call simulated H323 and RTP endpoints.
Table 4-105. Make_Call Activity Test Settings
Category Settings
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VH_027_GK_H323v4_NC_Basic_Call_with_HwRTP
This test is similar to the previous VH_019_GK_H323v4_NC_Basic_Call_with_RTP test, with the difference that the Enable Hw Acceleration option is selected in the H323 page of both origi-nating and terminating activities.
VH_028_GK_H323v4_NC_Basic_Call_with_HwRTP_10GA_ER_Trunk
This test is similar to the previous VH_027_GK_H323v4_NC_Basic_Call_with_HwRTP test, with the difference that the Allow multiple aggregated 1G ports option is selected in the Test Options configuration page.
Note: The rest of the H323 samples, ranging from VH_029 to VH_039, represent implementations of some of the previous test using IPv6 instead of IPv4 network level settings.
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H.248 Sample Test Configurations and Test Scenarios
This section describes the pre-defined IxLoad Voice Plug-in H.248/MEGACO available sample test configurations (RXFs) and their associated test scenarios.
Used Test Configurations
Most of the sample H248 tests supplied with IxLoad are based on one of the fol-lowing two configurations:
• MGW/MGC: IxLoad simulates a single MGW and its controlling MGC. The simulated H248 protocol and RTP functions flow is mapped to H248TermGroups that are associated with each of the H248MGW1 and H248MGW1 activities (Figure 4-83).
Figure 4-83. Single MGW with Controlling MGC
• Two MGWs/MGC: IxLoad simulates two MGW and their controlling MGC. The simulated H248 protocol and RTP functions flow is mapped to H248TermGroups that are associated with each of the H248MGW1, H248MGW1, and H248MGC1 activities. Please note that the H248MGC1 activity has two configured H248TermGroups, each one corresponding to a controlled GW (Figure 4-84).
Figure 4-84. Two MGWs with Controlling MGC
Note: For a complete description of the supported H.248 Test Library functions, refer to VoIP H248 Functions Library on page 3-123.
Note: Sample tests follow a naming convention that comprises test type (VM for VoIP MEGACO), an index, a test topology (T1 shown in Figure 4-83 or T2 shown in Figure 4-84), an IP protocol version (IPv4 or IPv6) a configuration (B2B or vs_DUT), a and a short call description, such as for example VM_001_H248_IPv4_B2B_with_version_3.
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VoIP H.248 Test Configurations
The following sample VoIP H.248 test configuration files are contained in the IxLoad installation kit:
VM_001_H248_IPv4_B2B_with_version_v3
This test based on the configuration shown in Figure 4-83 simulates an Access Gateway, with a configured objective of 2 channels, and a controlling MGC. The GW activity has a TermGroup defined that executes an H.248 signaling and media functions flow. The MGC has an associated TermGroup that executes an H248 and media functions flow.
The underlying two-channel test scenario involving H248TermGroupMGC1 and H248TermGroupMGW1 is shown in Figure 4-85.
Figure 4-85. VM_001_H248_IPv4_B2B_with_version_v3 Test Scenario
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-106.
Table 4-106. H248MGC1 Activity Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Access Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Notify requests, Send Modify on Root termination to set properties, Use TransactionResponseAck, and Wait for MGW registration options are selected.
Profiles The ETSI_ARGW/1 profile is selected.
H248TermGroupMGC1
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-107.
Scenario The test scenario comprises the Add and Modify functions followed by the VoiceSession function for media exchange. The final Subtract function performs the deletion of all active terminations.
Execution Settings The corresponding scenario channel is configured to execute twice during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-107. H248MGW1 Activity Test Settings
Category Settings
H248MGW1
Simulated MGW The simulated GW type is configured to Access Gateway and the controlling MGC is set to the H248MGC1 activity.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Audit requests, Auto-reply to Modify on Root termination, Use TransactionResponseAck, and the Auto-register options are selected.
Profiles The ETSI_ARGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Scenario The test scenario comprises the Wait Add, Wait Modify messages followed by a media session using the VoiceSession function. The last function is a Wait Subtract function.
Execution Settings The corresponding scenario channel is configured to execute twice during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-106. H248MGC1 Activity Test Settings
Category Settings
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VM_002_H248_IPv4_B2B_with_version_v2
This test is similar with that described in VM_001_H248_IPv4_B2B_with_version_v3, except that it uses version 2 of the H.248/MEGACO protocol.
VM_003_H248_IPv4_B2B_with_version_v1
This test is similar with that described in VM_001_H248_IPv4_B2B_with_version_v3, except that it uses version 1 of the H.248/MEGACO protocol.
VM_004_H248_IPv6_B2B_with_version_v3
This test is similar with that described in VM_001_H248_IPv4_B2B_with_version_v3, except that it uses IPv6 addressing instead of IPv4.
VM_005_H248_IPv6_B2B_with version_v2
This test is similar with that described in VM_002_H248_IPv4_B2B_with_version_v2, except that it uses IPv6 addressing instead of IPv4.
VM_006_H248_IPv6_B2B_with_version_v1
This test is similar with that described in VM_003_H248_IPv4_B2B_with_version_v1, except that it uses IPv6 addressing instead of IPv4.
VM_007_H248_IPv4_B2B_with_message_maximum_size_4000
This test is similar with that described in VM_001_H248_IPv4_B2B_with_version_v3, except that it uses a maximum H.248 message size of 4000 bytes.
VM_008_H248_IPv4_B2B_reply_send_AuditValue
This test based on the configuration shown in Figure 4-83 simulates a Trunking GW with a configured objective of 100 channels and a controlling MGC. The MGW activity has a TermGroup defined that executes an H.248 signaling and media functions flow. The MGC has an associated TermGroup that executes an H248 and media functions flow.
The underlying two-channel test scenario involving H248TermGroupMGC1 and H248TermGroupMGW1 is shown in Figure 4-87.
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Figure 4-86. VM_008_H248_IPv4_B2B_reply_send_AuditValue Test Scenario
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-108.
Table 4-108. H248MGC1 Activity Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Trunking Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Notify requests, Send Modify on Root termination to set properties, and Use TransactionResponseAck, options are selected.
The Wait for MGW Registration and the Send AuditValue options are also selected.
Profiles The ETSI_TGW/1 profile is selected.
H248TermGroupMGC1
Scenario The test scenario comprises the Add and Modify (auto SDP) functions followed by the VoiceSession function for media exchange. Eventually a Subtract function performs the deletion of all active terminations.
Execution Settings The corresponding scenario channel is configured to execute five times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-109.
VM_009_H248_IPv4_B2B_reply_send_AuditCapabilities
This test is similar with that described in VM_008_H248_IPv4_B2B_reply_send_AuditValue, except that it has the Send AuditCapabilities instead of the Send AuditValue option is configured in the Automatic Functionality page of the H248MGC1 activity.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-109. H248MGW1 Activity Test Settings
Category Settings
H248MGW1
Simulated MGW The simulated GW type is configured to Trunking Gateway and the controlling MGC is set to the H248MGC1 activity.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Audit requests, Auto-reply to Modify on Root termination, Use TransactionResponseAck, and the Auto-register options are selected.
Profiles The ETSI_TGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Scenario The test scenario comprises the Wait Add, Wait Modify messages followed by a media session using the VoiceSession function. the final function is a Wait Subtract function.
Execution Settings The corresponding scenario channel is configured to execute five times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-108. H248MGC1 Activity Test Settings
Category Settings
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VM_010_H248_IPv4_B2B_enable_retransmissions
This test is similar with that described in VM_009_H248_IPv4_B2B_reply_send_AuditCapabilities, except that it has the Enable Retransmissions option configured on the H248MGC1 activity (Auto-matic Functionality page).
At test scenario level, the Wait Add script function on the H248MGW1 activity has a configured delay of 600 ms, such as to enforce the retransmission of MGC Add messages.
VM_011_H248_IPv4_B2B_Access_gw
This test based on the configuration shown in Figure 4-83 simulates an Access GW with a configured objective of 100 channels and a controlling MGC. The GW activity has a TermGroup defined that executes an H.248 signaling and media functions flow. The MGC has an associated TermGroup that executes an H248 and media functions flow.
The underlying two-channel test scenario involving H248TermGroupMGC1 and H248TermGroupMGW1 is shown in Figure 4-87.
Figure 4-87. VM_011_H248_IPv4_B2B_Access_gw Test Scenario
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-110.
Table 4-110. H248MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Access Gateway (PSTN2IP) and H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-111.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Notify requests, Send Modify on Root termination to set properties, Use TransactionResponseAck, and Wait for MGW Registration options are selected.
Profiles The ETSI_ARGW/1 profile is selected.
H248TermGroupMGC1
Scenario Editor The test scenario comprises the Add and Modify (auto SDP descriptor) functions followed by the VoiceSession function for media exchange. The final Subtract function performs the deletion of all active terminations.
Execution Settings The corresponding scenario channel is configured to execute five times during the test sustain time.
SDP The auto SDP option at script function level is not overridden by any setting in this tab.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-111. H248MGW1 Activity Test Settings
Category Settings
H248MGW1
Simulated MGW The simulated GW type is configured to Access Gateway and the controlling MGC is set to the H248MGC1 activity.
H248TermGroupMGW1 is configured on the H248MGW1 activity.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Audit requests, Auto-reply to Modify on Root termination, Use TransactionResponseAck, and the Auto-register options are selected.
Profiles The ETSI_ARGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Scenario The test scenario comprises the Wait Add, Wait Modify messages followed by a media session that uses the VoiceSession script function. The last function is a Wait Subtract function.
Table 4-110. H248MGC1 Test Settings
Category Settings
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VM_012_H248_IPv4_B2B_Border_gw
This test based on the configuration shown in Figure 4-83 simulates a Border GW (IP2IP) with a configured objective of 10 channels, a controlling MGC, and four media-only VoIPSIPPeer activities that simulate RTP endpoints.
The MGC has an associated H248TermGroup that executes an H248-only func-tions flow. An RTP endpoints pair defined using two media-only VoIPSIPPeer activities simulate the RTP terminations associated with the MGC.
The MGW activity has an associated H248TermGroup that executes an H.248-only functions flow. An RTP endpoints pair defined using another two media-only VoIPSIPPeer activities simulate the RTP terminations associated with the MGW.
The media-only VoIPSIPPeer activities VoIPSIPPeer1 to VoIPSIPPeer4 are defined on four NetTraffics, each with a different underlying network range, and simulate RTP endpoints. Each RTP endpoints pair, VoIPSIPPeer2 – VoIPSIPPeer3 and VoIPSIPPeer1 – VoIPSIPPeer4 exchange media bidirection-ally.
The test configuration has two associated test scenarios:
• The first scenario VM_012_H248_IPv4_B2B_Border_gw_MGC comprises five channels:
• Channel#0: Executes an MGC H248 functions flow.
• Channel#1 to channel#4: Each channel executes a bidirectional media exchange using a VoiceSession script function.
• The second senario VM_012_H248_IPv4_B2B_Border_gw_MGW comprises a single channel that executes an H.248 functions flow mirroring that from channel#0 of the VM_012_H248_IPv4_B2B_Border_gw_MGC test scenario.
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-112.
Execution Settings The corresponding scenario channel is configured to execute five times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-111. H248MGW1 Activity Test Settings
Category Settings
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Table 4-112. H248MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Border Gateway (IP2IP) and H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Wait for Registration option is selected.
Profiles The ETSI_GateControl/1 profile is selected.
H248TermGroupMGC1
Scenario The activity is linked to the first channel of the VM_012_H248_IPv4_B2B_Border_gw_MGC test scenario comprising the following script functions:
• Sleep
• Add
• Modify: Conveys to the first RTP termination a remoteDescriptor with a custom SDP definition based on the VOIPVar0 and VOIP_IPAddress0 variables.
Conveys to the second RTP termination a remoteDescriptor with a custom SDP definition based on the VOIPVar1 and VOIP_IPAddress1 variables.
• Subtract
• Sleep
Execution Settings The corresponding scenario channel is configured to execute three times during the test sustain time.
SDP No overriding SDP settings are used. The Skip SDP processing option is not selected.
Codec Settings The default G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is not selected.
Other Settings The scenario variables are used as follows:
• VOIP_IPAddress0 and VOIP_Var0 are initialized to the IP address and port values of the VoIPSIPPeer3-emu-lated RTP endpoints.
• VOIP_IPAddress1 and VOIP_Var1 are initialized to the IP address and port values of the VoIPSIPPeer1-emu-lated RTP endpoints.
• VOIP_IPAddress2 and VOIP_Var2 are initialized to the IP address and port values of the VoIPSIPPeer2-emu-lated RTP endpoints.
• VOIP_IPAddress2 and VOIP_Var2 are initialized to the IP address and port values of the VoIPSIPPeer4-emu-lated RTP endpoints.
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-113.
Table 4-113. H248MGW1 Test Settings
Category Settings
H248MGW1
Simulated MGC The controlled GW type is configured to Border Gateway (IP2IP) and H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto Register option is selected.
Profiles The ETSI_GateControl/1 profile is selected.
H248TermGroupMGW1
Scenario The activity is linked to the first channel of the VM_012_H248_IPv4_B2B_Border_gw_MGW test scenario comprising the following script functions:
• Sleep
• Wait Add: For the first RTP termination it returns a localDescriptor with a custom SDP definition, based on the VOIPVar2 and VOIP_IPAddress2 variables confi-gured in the H248MGW1 activity.
For the second RTP termination it returns localDescriptor with a custom SDP definition, based on the VOIPVar3 and VOIP_IPAddress3 variables configured in the H248MGW1 activity.
• Modify
• Wait Subtract
• Sleep
Execution Settings The corresponding scenario channel is configured to execute three times during the test sustain time.
SDP No overriding SDP settings are used.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
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The configuration of all four simulated VoIPSIPPeer activities is similar to that of the VoIPSIPPeer1 activity, described in Table 4-114.
VM_013_H248_IPv4_B2B_Trunking_gw
This test is similar with that described in VM_011_H248_IPv4_B2B_Access_gw, except that it simulates a Trunking Gateway instead of an Access Gateway.
RTP Settings The Enable media on this activity option is not selected.
Other Settings The scenario variables are used as follows:
• VOIP_IPAddress0 and VOIP_Var0 are initialized to the IP address and port values of the VoIPSIPPeer3-emu-lated RTP endpoints.
• VOIP_IPAddress1 and VOIP_Var1 are initialized to the IP address and port values of the VoIPSIPPeer1-emu-lated RTP endpoints.
• VOIP_IPAddress2 and VOIP_Var2 are initialized to the IP address and port values of the VoIPSIPPeer2-emu-lated RTP endpoints.
• VOIP_IPAddress2 and VOIP_Var2 are initialized to the IP address and port values of the VoIPSIPPeer4-emu-lated RTP endpoints.
Table 4-114. VoIPSIPPeer1 Activity Test Settings
Category Settings
Scenario The underlying channel comprises a single VoiceSession function for bidirectional media exchange.
Execution Settings The following RTP Channel Mapping settings are used:
• IP: Use Consecutive values (per port)
• UDP port: Use consecutive values (per port)
Dial Plan The VoIPSIPPeer4 : <portrange> activity is configured as call destination. The portrange value is that configured for the VOIP_Var3 variable in the Other Settings tab of the H248MGC1 activity.
SIP Settings The use of signaling is not selected.
Codec Settings The default codec settings are used.
RTP Settings The Enable media on this activity option is selected and the RTP port is defined using a sequence generator expression.
Other Settings The IP version preference is set to IPv4, and no scenario variables need to be initialized.
Table 4-113. H248MGW1 Test Settings
Category Settings
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VM_014_H248_IPv4_B2B_Residential_gw
This test based on the configuration shown in Figure 4-84 simulates two Resi-dential GWs, each with a with a configured objective of 10 channels, and a con-trolling MGC.
Each MGW activity has an H248TermGroup defined that executes an H.248 and media functions flow. The MGC has two associated TermGroups that execute an H248-only functions flow.
The underlying four-channel test scenario involving H248TermGroupMGW1, H248TermGroupMGW2, H248TermGroupMGC1, and H248TermGroupMGC2 is shown in Figure 4-88.
Figure 4-88. VM_014_H248_IPv4_B2B_Residential_gw Test Scenario
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-119.
Table 4-115. MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Residential Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled.
H248 Version 1 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Wait for MGW Registration option is selected.
Profiles The ETSI_TGW/1 profile is selected.
H248TermGroupMGC1
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-120.
Scenario The test scenario channel comprises the following functions:
• Modify
• Wait Notify
• Modify
• Wait Notify
• Add: the received localDescriptor value is stored into the RemoteDescriptor variable.
• Modify: the sent remoteDescriptor has a custom defini-tion that uses the $RemoteDescriptor[$arrScenar-ioCh[2]] variable corresponding to the SDP value from the scenario channel #2.
• Wait Notify
• Subtract
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time.
SDP The Skip SDP processing option is selected.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is not selected.
Other Settings No scenario variables need to be initialized.
Table 4-116. MGW1 Settings
Category Settings
H248MGW1
Simulated MGW The simulated GW type is configured to Residential Gateway and the controlling MGC is set to the H248MGC1 activity. H248TermGroupMGW1 is configured on the MGW activity.
H248 Version 1 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to Modify on Root termination and Auto Register options are selected.
Profiles The ETSI_TGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Table 4-115. MGC1 Test Settings
Category Settings
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VM_015_H248_IPv4_B2B_MID_IP_Address
This test based on the configuration shown in Figure 4-83 simulates a Trunking Gateway, with a configured objective of 10 channels, and a controlling MGC.
Both the H248MGW1 and H248MGC1 activity have H248TermGroups defined that execute an H.248 and RTP functions flow.
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-117.
Scenario The test scenario channel comprises the following functions:
• Wait Modify
• Notify
• Wait Modify
• Notify
• Wait Add
• Wait Modify
• VoiceSession: Performs bidirectional media exchange
• Notify: Notifies the MGC of an onhook condition
• Wait Subtract
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Note: H248MGW2 and H248TermGroupMGW2 are configured similar to H248MGW1 and H248TermGroupMGW1 respectively, with the exception of the underlying test scenario channel. The Add function stores the received localDescriptor value in the RemoteDescriptor variable and sends a custom remoteDescriptor based on the $RemoteDescriptor[$arrScenarioCh[1]] variable corresponding to the SDP value from scenario channel #1.
H248TermGroupMGC2 is configured similar to H248TermGroupMGC1.
Table 4-116. MGW1 Settings
Category Settings
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-118.
Table 4-117. MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The MID format parameter is configured to the IP Address setting.
The controlled GW type is configured to Trunking Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Notify requests, Send Modify on Root termination to set properties, Use TransactionResponseAck, and Wait for MGW registration options are selected.
Profiles The ETSI_TGW/1 profile is selected.
H248TermGroupMGC1
Scenario The test scenario channel comprises an Add and a Modify function. Media exchange is performed using a VoiceSession function. Eventually the terminations are disconnected using a Subtract function.
Execution Settings The corresponding scenario channel is configured to execute three times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-118. MGW1 Settings
Category Settings
H248MGW1
Simulated MGW The MID format parameter is configured to the IP address setting.
The simulated GW type is configured to Trunking Gateway and the controlling MGC is set to the H248MGC1 activity.
H248TermGroupMGW1 is configured on the MGW activity.
H248 Version 3 of the H.248/MEGACO protocol is configured.
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VM_016_H248_IPv4_B2B_MID_Device_Name
This test is similar to VM_015_H248_IPv4_B2B_MID_IP_Address, except that it uses a Device name option for the MID parameter.
VM_017_H248_IPv4_B2B_MID_IPAddress_Port
This test is similar to VM_015_H248_IPv4_B2B_MID_IP_Address, except that it uses an IP Address: Port option for the MID parameter.
VM_018_H248_IPv4_B2B_MID_MGW_MGC_DNS_Name
This test is similar to VM_015_H248_IPv4_B2B_MID_IP_Address, except that it uses a MGC DNS name / MGW DNS name option for the MID parameter.
VM_019_H248_IPv4_B2B_performance_topology2_rtp
This test based on the configuration shown in Figure 4-84 simulates two Resi-dential GWs, each with a with a configured objective of 900 channels, and a con-trolling MGC.
Each MGW activity has an H248TermGroup defined that executes an H.248 and media functions flow. The MGC has two associated TermGroups that execute an H248-only functions flow.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Audit requests, Auto-reply to Modify on Root termination, Use TransactionResponseAck, and the Auto-register options are selected.
Profiles The ETSI_TGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Scenario The test scenario channel comprises the script functions corresponding to the MGC-transmitted commands (Table 4-121).
Execution Settings The corresponding scenario channel is configured to execute three times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-118. MGW1 Settings
Category Settings
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The underlying four-channel test scenario involving H248TermGroupMGW1, H248TermGroupMGW2, H248TermGroupMGC1, and H248TermGroupMGC2 is shown in Figure 4-89.
Figure 4-89. VM_019_H248_IPv4_B2B_performance_topology2_rtp Test Scenario
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-119.
Table 4-119. MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Residential Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Wait for MGW Registration option is selected.
Profiles The ETSI_TGW/1 profile is selected.
H248TermGroupMGC1
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-120.
Scenario The test scenario channel comprises the following functions:
• Modify
• Wait Notify
• Modify
• Wait Notify
• Add: the received localDescriptor value is stored into the RemoteDescriptor variable.
• Modify: the sent remoteDescriptor has a custom defini-tion that uses the $RemoteDescriptor[$arrScenar-ioCh[2]] variable corresponding to the SDP value from the scenario channel #2.
• Wait Notify
• Subtract
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time.
SDP The Skip SDP processing option is selected.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is not selected.
Other Settings No scenario variables need to be initialized.
Table 4-120. MGW1 Settings
Category Settings
H248MGW1
Simulated MGW The simulated GW type is configured to Residential Gateway and the controlling MGC is set to the H248MGC1 activity. H248TermGroupMGW1 is configured on the MGW activity.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to Modify on Root termination and Auto Register options are selected.
Profiles The ETSI_TGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Table 4-119. MGC1 Test Settings
Category Settings
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VM_020_H248_IPv4_B2B_performance_topology1_rtp
This test based on the configuration shown in Figure 4-83 simulates a Trunking GW, with a configured objective of 900 channels, and a controlling MGC.
Both the H248MGW1 and H248MGC1 activity have H248TermGroups defined that execute an H.248 and media functions flow.
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-121.
Scenario The test scenario channel comprises the following functions:
• Wait Modify
• Notify
• Wait Modify
• Notify
• Wait Add
• Wait Modify
• VoiceSession: Performs bidirectional media exchange
• Notify: Notifies the MGC of an onhook condition
• Wait Subtract
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Note: H248MGW2 and H248TermGroupMGW2 are configured similar to H248MGW1 and H248TermGroupMGW1 respectively, with the exception of the underlying test scenario channel. The Add function stores the received localDescriptor value in the RemoteDescriptor variable and sends a custom remoteDescriptor based on the $RemoteDescriptor[$arrScenarioCh[1]] variable corresponding to the SDP value from scenario channel #1.
H248TermGroupMGC2 is configured similar to H248TermGroupMGC1.
Table 4-120. MGW1 Settings
Category Settings
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-122.
Table 4-121. MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Trunking Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Notify requests, Send Modify on Root termination to set properties, Use TransactionResponseAck, and Wait for MGW registration options are selected.
Profiles The ETSI_TGW/1 profile is selected.
H248TermGroupMGC1
Scenario The test scenario channel comprises two Modify functions for the physical terminations, followed by an Add and a Modify function for the RTP terminations. Media is exchanged bidirectionally using a VoiceSession function. Eventually the terminations are disconnected using a Subtract function.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-122. MGW1 Settings
Category Settings
H248MGW1
Simulated MGW The simulated GW type is configured to Trunking Gateway and the controlling MGC is set to the H248MGC1 activity.
H248TermGroupMGW1 is configured on the MGW activity.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Audit requests, Auto-reply to Modify on Root termination, Use TransactionResponseAck, and Auto-register options are selected.
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VM_021_H248_IPv4_B2B_without_registration
This test is similar to VM_020_H248_IPv4_B2B_performance_topology1_rtp, except that no registration of the MGW is performed with the MGC. As such, the test does not have the Wait for MGW Registration (on the H248MGC1 activ-ity) and the the Auto Register (on the H248MGW1 activity) options configured.
VM_022_H248_IPv4_B2B_G711_ulaw
This test based on the configuration shown in Figure 4-83 simulates a Trunking GW with a configured objective of 20 channels and a controlling MGC. The GW activity has an H248TermGroup defined that executes an H.248 signaling and media functions flow. The MGC has an associated H248TermGroup that exe-cutes an H248 and media functions flow.
The underlying two channel test scenario involving H248TermGroupMGC1 and H248TermGroupMGW1 is shown in Figure 4-90.
Profiles The ETSI_TGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Scenario The test scenario channel comprises the script functions corresponding to the MGC transmitted commands (Table 4-121). Media is exchanged bidirectionally using a VoiceSession function.
Execution Settings The corresponding scenario channel is configured to execute repeatedly during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-122. MGW1 Settings
Category Settings
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Figure 4-90. VM_022_H248_IPv4_B2B_G711_ulaw Test Scenario
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-123.
Table 4-123. MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Trunking Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Notify requests, Send Modify on Root termination to set properties, Use TransactionResponseAck, and Wait for MGW registration options are selected.
Profiles The ETSI_TGW/1 profile is selected.
H248TermGroupMGC1
Scenario The test scenario comprises the Add, Modify messages followed by a media session that uses the VoiceSession script function. The terminations are eventually deleted using a Subtract function.
Execution Settings The corresponding scenario channel is configured to execute five times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 u-law and G.711 a-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-124.
VM_023_H248_IPv4_B2B_G711_alaw
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the G711 alaw instead of the G711 ulaw codec.
VM_024_H248_IPv4_B2B_G723_1_5.3
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the G723 @5.3kbps instead of the G711 ulaw codec.
VM_025_H248_IPv4_B2B_G723_1_6.3
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the G723 @6.3kbps instead of the G711 ulaw codec.
Table 4-124. MGW1 Settings
Category Settings
H248MGW1
Simulated MGW The simulated GW type is configured to Trunking Gateway and the controlling MGC is set to the H248MGC1 activity.
H248TermGroupMGW1 is configured and enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Audit requests, Auto-reply to Modify on Root termination, Use TransactionResponseAck, and Auto-register options are selected.
Profiles The ETSI_TGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Scenario The test scenario comprises the Wait Add, Wait Modify, followed by a media session that uses the VoiceSession script function, and the Wait Subtract function.
Execution Settings The corresponding scenario channel is configured to execute five times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 u-law and G.711 a-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
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VM_026_H248_IPv4_B2B_G726_16
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the G726 @16kbps instead of the G711 ulaw codec.
VM_027_H248_IPv4_B2B_G726_24
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the G726 @16kbps instead of the G711 ulaw codec.
VM_028_H248_IPv4_B2B_G726_32
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the G726 @32kbps instead of the G711 ulaw codec.
VM_029_H248_IPv4_B2B_G726_40
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the G726 @40kbps instead of the G711 ulaw codec.
VM_030_H248_IPv4_B2B_G729AB
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the G729AB instead of the G711 ulaw codec.
VM_031_H248_IPv4_B2B_ILBC
This test is similar to VM_022_H248_IPv4_B2B_G711_ulaw, except that it uses the iLBC instead of the G711 ulaw codec.
VM_032_H248_IPv4_B2B_auto_sdp
This test based on the configuration shown in Figure 4-83 simulates a Trunking GW with a configured objective of 20 channels and a controlling MGC. The GW activity has an H248TermGroup defined that executes an H.248 signaling and media functions flow. The MGC has an associated H248TermGroup that exe-cutes an H248 and media functions flow.
The underlying two channel test scenario involving H248TermGroupMGC1 and H248TermGroupMGW1 is shown in Figure 4-91.
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Figure 4-91. VM_032_H248_IPv4_B2B_auto_sdp Test Scenario
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-125.
Table 4-125. MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Trunking Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Notify requests, Send Modify on Root termination to set properties, Use TransactionResponseAck, and Wait for MGW registration options are selected.
Profiles The ETSI_TGW/1 profile is selected.
H248TermGroupMGC1
Scenario The test scenario comprises the Add (localDescriptor = AutoSDP), Modify (RemoteDescriptor = AutoSDP) messages followed by a media session that uses the VoiceSession script function. The terminations are eventually deleted using a Subtract function.
Execution Settings The corresponding scenario channel is configured to execute four times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 u-law, G.711 u-law, G.726@16kbps, and G.726@24kbps codecs are selected.
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The H248MGW1 and H248TermGroupMGW1 configured settings are described in Table 4-126.
VM_033_H248_IPv4_B2B_custom_SDP
This test is similar to VM_032_H248_IPv4_B2B_auto_sdp, except that it uses the a custom SDP definition in the Modify script function on scenario channel #0.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-126. MGW1 Settings
Category Settings
H248MGW1
Simulated MGW The simulated GW type is configured to Trunking Gateway and the controlling MGC is set to the H248MGC1 activity.
H248TermGroupMGW1 is configured and enabled.
H248 Version 3 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto-reply to ServiceChange requests, Auto-reply to Audit requests, Auto-reply to Modify on Root termination, Use TransactionResponseAck, and the Auto-register options are selected.
Profiles The ETSI_TGW/1 profile is selected for the simulated MGW.
H248TermGroupMGW1
Scenario The test scenario comprises script functions matching the scenario channel#0 functions flow (Table 4-125).
Execution Settings The corresponding scenario channel is configured to execute four times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 u-law, G.711 u-law, G.726@16kbps, and G.726@24kbps codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-125. MGC1 Test Settings
Category Settings
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VM_034_H248_IPv4_B2B_sdp_renegociation
This test is similar with VM_033_H248_IPv4_B2B_custom_SDP, with the dif-ference that at test scenario level, after the first media exchange session, the codec is re-negotiated using another Modify function. Eventually media is exchanged again using a VoiceSession function.
VM_035_H248_IPv4_vs_DUT_RGW_analog_basic_call
This test simulates an MGC activity and runs against a Residential GW (DUT).
The H248MGC1 and H248TermGroupMGC1 configured settings are described in Table 4-127.
VM_036_H248_IPv4_vs_DUT_RGW_analog_basic_all_with_renegotiation
This test is similar to VM_035_H248_IPv4_vs_DUT_RGW_analog_basic_call, except that codec renegotiation is done at scenario level using Modify messages. After each negotiation, media is exchanged using VoiceSession script functions.
Table 4-127. MGC1 Test Settings
Category Settings
H248MGC1
Simulated MGC The controlled GW type is configured to Residential Gateway (PSTN2IP) and the H248TermGroupMGC1 is enabled. The IP address of the controlled GW is configured in the MGW column.
H248 Version 1 of the H.248/MEGACO protocol is configured.
Automatic Functionality
The Auto reply to ServiceChange requests and Wait for MGW registration options are selected.
Profiles The ETSI_ARGW/1 profile is selected.
H248TermGroupMGC1
Scenario The test scenario comprises an initial Modify for the physical termination, followed by Add, Modify functions for the RTP terminations. Media is exchanged using the VoiceSession script function and the terminations are eventually deleted using a Subtract function.
Execution Settings The corresponding scenario channel is configured to execute five times during the test sustain time.
SDP No overriding SDP settings are configured.
Codec Settings The G.711 a-law and G.711 u-law codecs are selected.
RTP Settings The Enable media on this activity option is selected.
Other Settings No scenario variables need to be initialized.
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VM_037_H248_IPv4_B2B_QoV
This test is similar to VM_032_H248_IPv4_B2B_auto_sdp, except that QoV computation is selected in the RTP page of both H248TermGroup activities.
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MGCP Sample Test Configurations and Test Scenarios
This section describes the predefined IxLoad Voice Plug-in MGCP available sample test configurations (RXFs) and their associated test scenarios.
Used Test Configurations
All sample MGCP tests supplied with IxLoad are based on one of the following two configurations:
• GW/GW: IxLoad is used to simulate two GWs, while the CA is a real test device (DUT). The combined MGCP protocol and RTP functions flow is mapped to Endpoints that are associated with each of the MGCP GW activi-ties (Figure 4-92).
Figure 4-92. Two GW with Real CA (DUT)
• GW/CA: IxLoad is used to simulate a GW and its controlling CA. In this test configuration, the CA is intended to simulate an additional GW that is con-trolled by the CA and whose endpoints perform RTP media exchange with the MGCPGW1 endpoints.
The combined MGCP protocol and RTP functions flow is mapped to End-points that are associated with each of the MGCP GW and MGCP CA activi-ties (Figure 4-93).
Figure 4-93. GW with Controlling CA
• Two GWs/CA: IxLoad is used to simulate two GWs and their controlling CA. The combined MGCP protocol and RTP functions flow is mapped to Endpoints that are associated with each of the MGCPGW1 and MGCPGW2 activities. Please note that the MGCPCA1 activity is configured with two Endpoint activities, each one corresponding to a controlled GW (Figure 4-94), each one executing an MGCP-only protocol flow.
Note: For a complete description of the supported H.248 Test Library functions, refer to VoIP MGCP Functions Library on page 3-138.
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Figure 4-94. Two GWs with Controlling CA
VoIP MGCP Test Configurations
The following sample VoIP MGCP test configuration files are contained in the IxLoad installer:
VMG_001_MGCP_IPV4_B2B_Basic_Call_Gw_vs_CA_with_RTP
This test based on the configuration shown in Figure 4-93 runs in B2B mode and simulates MGCPGW1 endpoints calling other endpoints, with media streaming performed after call setup.
Both the GW and the CA activity have Endpoints defined that execute an MGCP signaling and RTP media functions flow.
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-128.
Table 4-128. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 100 gateways (gw[001-100]) are configured with a single endpoint each. The controlling CA is MGCPCA1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint1
Scenario The scenario channel implements a simple call sequence with media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The destination phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
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The MGCPCA1 and Endpoint2 configured settings are described in Table 4-129.
VMG_002_MGCP_IPV4_B2B_Basic_Call_Gw_vs_CA_signalling_only
This test is similar to VMG_001_MGCP_IPV4_B2B_Basic_Call_Gw_vs_CA_ with_RTP, with the only difference that the call flow is signaling-only (no Voice Session functions).
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-129. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
MGCPGCA1
Simulated Gateways
The controlled gateways are specified as one set with a number of 100 gateways (gw[001-100]) with an endpoint each, corresponding to MGCPGW1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint2
Scenario The scenario channel implements a simple call sequence (receiving side) with media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-128. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
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VMG_003_MGCP_IPV4_B2B_Basic_Call_GW1_calls_GW2_through_CA
This test based on the configuration shown in Figure 4-94 test runs in B2B mode and simulates GW1 endpoints calling endpoints provisioned on another GW via a CA, with media streaming performed after call setup.
Both GWs and the CA activity have Endpoints defined that execute an MGCP signaling flow (between the GW and CA) and RTP media functions flow (between endpoints on GWs).
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-130.
Table 4-130. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 8000 gateways (gw[00000-07999]) are configured with a single endpoint each. The controlling CA is MGCPCA1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint1
Scenario Endpoints perform an initialization procedure and initiate the call by exchanging MGCP messages with the CA, after which media streaming is performed using the VoiceSession script function. Eventually the endpoints terminate the call following a request from the CA.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The destination phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP No RTP settings are configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
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The MGCPCA1 and Endpoint2 configured settings are described in Table 4-131.
Note: The scenario-level configuration for MGCPGW2 and Endpoint3 is similar to the previous one, with the difference that the Endpoint3-simulated endpoints initiate the call termination.
At activity-level, MGCPGW2 is configured to simulate number of 8000 gateways (gw[08000-15999]) with a single endpoint each. On Endpoint 3 the call destination is configured as 160[00000-], but it is not used by the test, since Endpoint3 only receives a call.
Table 4-131. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
MGCPCA1
Simulated Call Agent
The controlled gateways are specified as two sets, one with a number of 8000 gateways corresponding to MGCPGW1 and another set of 8000 gateways corresponding to MGCPGW2.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option is selected.
Endpoint2
Scenario The test scenario implements the CA-side MGCP-only message exchange with the endpoints simulated by MGCPGW1.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The source phone number for MGCPGW1 endpoints is specified using the 160[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP No RTP settings are configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Note: The Endpoint4 activity implements the CA-side MGCP-only message exchange with MGCPGW2 simulated endpoints. Endpoint4 is configured similar to Endpoint2, with the difference that it specifies the source phone numbers using the 170[00000-] sequence generator expression.
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VMG_004_MGCP_IPV4_B2B_Basic_Call_with_Renegociation
This test based on the configuration shown in Figure 4-93 runs in B2B mode and simulates MGCPGW1 endpoints calling other endpoints, with media streaming performed after call setup. Following a first media streaming session, codec rene-gotiation occurs (MDCX) and another media streaming session is performed.
Both the GW and the CA activity configured by this have Endpoint activities associated defined that execute an MGCP signaling and RTP media functions flow.
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-132.
The MGCPCA1 and Endpoint2 configured settings are described in Table 4-133.
Table 4-132. MGCPGW1 Activity Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 100 gateways (gw[001-100]) are configured with a single endpoint each. The controlling CA is MGCPCA1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint1
Scenario The test scenario comprises the GW Make Call procedure, Voice Session function, Wait MDCX, Voice Session, and GW End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute 1 loop during the test sustain time.
Simulated Endpoints
The destination phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
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VMG_005_MGCP_IPV4_B2B_Basic_Call_with_DTMFs
This test is similar to VMG_001_MGCP_IPV4_B2B_Basic_Call_Gw_vs_CA_ with_RTP, with the main difference that the underlying test scenario contains Generate DTMF / Detect DTMF functions instead of the Voice Session func-tions.
The test runs in B2B mode and simulates MGCPGW1 endpoints calling other endpoints, with DTMF sending/receiving performed after call setup.
Both the GW and the CA activity have Endpoint activities defined that execute an MGCP signaling and a media functions flow.
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-134.
Table 4-133. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
MGCPGW2
Simulated Call Agent
The controlled gateways are specified as a set of 100 gateways (gw[001-100]) corresponding to MGCPGW1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint2
Scenario The test scenario mirrors the Endpoint1 call flow for the receiving side and comprises the Wait ReceiveCall, Voice Session function, Send MDCX, Voice Session, and Wait Recv_EndCall.
Execution Settings The corresponding scenario channel is configured to execute 1 loop during the test sustain time.
Simulated Endpoints
The endpoint phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
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The MGCPCA1 and Endpoint2 configured settings are described in Table 4-139.
Table 4-134. MGCPGW1 and Endpoint1 Activity Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 100 gateways (gw[001-100]) are configured with a single endpoint each. The controlling CA is MGCPCA1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option is selected.
Endpoint1
Scenario The test scenario comprises the GW Make Call procedure, Generate DTMF function, and GW End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The destination phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-135. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
MGCPGW2
Simulated Call Agent
The controlled gateways are specified as one set with a number of 100 gateways ([001-100]) corresponding to MGCPGW1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint2
Scenario The test scenario mirrors the Endpoint1 call flow for the receiving side and comprises the Wait ReceiveCall, Receive DTMF, and Wait Recv_EndCall.
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VMG_006_MGCP_IPV4_B2B_Basic_Call_with_voice_session_G711_Alaw
This test based on the configuration shown in Figure 4-93 runs in B2B mode and simulates MGCPGW1 endpoints calling other endpoints, with media streaming performed after call setup.
Both the GW and the CA activity configured by this test have Endpoint activities associated defined that execute an MGCP signaling and RTP media functions flow.
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-136.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The endpoint phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-136. MGCPGW1 and Endpoint1 Activity Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 100 gateways (gw[001-100]) are configured with a single endpoint each. The controlling CA is MGCPCA1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint1
Scenario The test scenario implements a common call setup sequence (caller) with media exchange following call establishment. It comprises the GW Make Call procedure, Voice Session function, and GW End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Table 4-135. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
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The MGCPCA1 and Endpoint2 configured settings are described in Table 4-139.
Simulated Endpoints
The destination phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-137. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
MGCPGW2
Simulated Call Agent
The controlled gateways are specified as a set of 100 gateways (gw[001-100]) corresponding to MGCPGW1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint2
Scenario The test scenario mirrors the Endpoint1 call flow for the receiving side (called party) and comprises the Wait ReceiveCall, Voice Session, and Wait Recv_EndCall.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The endpoint phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-136. MGCPGW1 and Endpoint1 Activity Settings
Category Settings
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VMG_007_MGCP_IPV4_B2B_Basic_Call_with_voice_session_G726-40
This test is similar to that VMG_006_MGCP_IPV4_B2B_Basic_Call_with_voice_ session_G711_Alaw, with the only difference that RTP streaming uses the G726@40kbps codec, con-figured in the Codecs tab of each Endpoint activity.
VMG_008_MGCP_IPV4_B2B_Basic_Call_with_RSIP_from_scenario
This test based on the configuration shown in Figure 4-93 runs in B2B mode and simulates MGCPGW1 endpoints calling other endpoints, with media streaming performed after call setup.
The main difference to VMG_009_MGCP_IPV4_B2B_Basic_Call_with_strip_ leading_zero_enabled is that at test scenario level MGCPGW1 endpoints are configured to send an MGCP RSIP message in the Init_Endpoint procedure that is executed prior to initiating the call setup sequence. Correspondingly, the CA-side message flow on the Endpoint2 activity contains in the Init_Endpoint pro-cedure a Wait RSIP script function that handles the incoming RSIP message.
VMG_009_MGCP_IPV4_B2B_Basic_Call_with_strip_leading_zero_enabled
This test based on the configuration shown in Figure 4-93 runs in B2B mode and simulates MGCPGW1 endpoints calling other endpoints, with RTP media streaming performed after call setup.
Both the GW and the CA activity configured by this test have Endpoint activities associated defined that execute an MGCP signaling and an RTP media functions flow.
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-138.
Table 4-138. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 100 simulated gateways (gw[001-100]) are configured with 900 endpoints (aaln[001-900]) each. The controlling CA is MGCPCA1.
The Strip leading zeros from endpoint name option is enabled.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint1
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The MGCPCA1 and Endpoint2 configured settings are described in Table 4-139.
Scenario The test scenario implements a common call setup sequence (caller) with media exchange following call establishment. It comprises the GW Make Call procedure, Voice Session function, and GW End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute 10 loops during the test sustain time.
Simulated Endpoints
The destination phone number is specified using the 170[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-139. MGCPCA1 and Endpoint 2 Activity Test Settings
Category Settings
MGCPGW2
Simulated Call Agent
The number of controlled gateways is specified one set of 100 gateways with 900 endpoints (aaln[001-900]) each, corresponding to MGCPGW1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint2
Scenario The test scenario mirrors the Endpoint1 call flow for the receiving side (called party) and comprises the Wait ReceiveCall, Voice Session, and Wait Recv_EndCall.
Execution Settings The corresponding scenario channel is configured to execute 10 loops during the test sustain time.
Simulated Endpoints
The endpoint phone number is specified using the 170[00000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
Table 4-138. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
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VMG_010_MGCP_IPV4_VS_DUT_BTS_Basic_Call_with_BTS
This test based on the configuration shown in Figure 4-92 runs against a real Call Agent device (BTS) and simulates MGCPGW1 endpoints calling other endpoints via the CA, with media streaming performed between endpoints after call setup.
Each MGW has an Endpoint activity defined that executes the MGCP flow with the CA and the media streaming towards the opposite MGW.
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-140.
The MGCPGW2 and Endpoint2 configured settings are described in Table 4-141.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-140. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 100 simulated gateways (ix[8002-8101]) are configured with a single endpoint each. The controlling CA is the DUT, which is specified by its IP address.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint1
Scenario MGCPGW1 endpoints call the endpoints on MGCPGW2 via the CA and perform media streaming (VoiceSession script function) over the established call.
Execution Settings The corresponding scenario channel is configured to execute repeatedly for the test sustain time.
Simulated Endpoints
The destination phone number is specified using a sequence generator expression which generates 100 numbers.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP No special RTP settings are configured.
Audio The Enable audio on this activity option is selected.
Other Settings No scenario variables need to be initialized.
Table 4-139. MGCPCA1 and Endpoint 2 Activity Test Settings
Category Settings
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VMG_011_MGCP_IPV4_B2B_HWRTP_Non_Agg_Basic_Call_RTP_8000ch
This test based on the configuration shown in Figure 4-93 runs in B2B mode and simulates endpoints calling other endpoints, with RTP media streaming per-formed after call setup.
Both the GW and the CA activity configured by this test have Endpoint activities associated defined that execute an MGCP signaling and an RTP media functions flow.
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-142.
Table 4-141. MGCPGW2 and Endpoint2 Activity Test Settings
Category Settings
MGCPGW2
Simulated Gateways
A number of 100 simulated gateways (ix[8102-8201]) are configured with a single endpoint each. The controlling CA is the DUT, is specified by its IP address.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option are selected.
Endpoint2
Scenario MGCPGW2 endpoints exchange MGCP messages with the CA for receiving the incoming call and then stream RTP media (VoiceSession script function) over the established call.
Execution Settings The corresponding scenario channel is configured to execute repeatedly for the test sustain time.
Simulated Endpoints
No destination phone number is specified.
SDP No overriding SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP No special RTP settings are configured.
Audio The Enable audio on this activity option is selected.
Other Settings No scenario variables need to be initialized.
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The MGCPCA1 and Endpoint2 configured settings are described in Table 4-143.
Table 4-142. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 8000 gateways (gw[0001-8000]) are configured with a single endpoint each. The controlling CA is MGCPCA1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option is selected.
Endpoint1
Scenario The test scenario implements a common call setup sequence (caller) with media exchange following call establishment. It comprises the GW Make Call procedure, Voice Session function, and GW End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The destination phone number is specified using the 160[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-143. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
MGCPGW2
Simulated Call Agent
The number of controlled gateways is specified one set of 8000 gateways corresponding to MGCPGW1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option is selected.
Endpoint2
Scenario The test scenario mirrors the Endpoint1 call flow for the receiving side (called party) and comprises the Wait ReceiveCall, Voice Session, and Wait Recv_EndCall.
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VMG_012_MGCP_IPV4_B2B_HWRTP_1G_Agg_Basic_Call_RTP_8000ch
This test is similar to VMG_011_MGCP_IPV4_B2B_HWRTP_Non_Agg_Basic_ Call_RTP_8000ch, with the only difference that 1 GB traffic aggregation for Acceleron load modules is used.
VMG_013_MGCP_IPV4_B2B_HWRTP_10G_Agg_Basic_Call_RTP_96000ch
This test based on the configuration shown in Figure 4-93 runs in B2B mode and simulates GW endpoints calling other endpoints, with RTP media streaming per-formed after call setup.
Both the GW and the CA activity configured by this test have Endpoint activities associated defined that execute an MGCP signaling and an RTP media functions flow.
The test has an associated test objective of 96000 channels and has 10 Gb traffic aggregation for Acceleron load modules configured.
The MGCPGW1 and Endpoint1 configured settings are described in Table 4-142.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The endpoint phone number is specified using the 160[00000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-144. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
MGCPGW1
Simulated Gateways
A number of 10 gateways (gw[01-12]) are configured with a number of 8000 (aaln[0001-8000]) endpoints each. The controlling CA is MGCPCA1.
Table 4-143. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
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The MGCPCA1 and Endpoint2 configured settings are described in Table 4-143.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option is selected.
Endpoint1
Scenario The test scenario implements a common call setup sequence (caller) with media exchange following call establishment. It comprises the GW Make Call procedure, Voice Session function, and GW End Call procedure.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The destination phone number is specified using the 160[000000-] sequence generator expression.
SDP No custom SDP settings are configured.
Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-145. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
MGCPGW2
Simulated Call Agent
The number of controlled gateways is specified one set of 12 gateways with 8000 endpoints each, corresponding to MGCPGW1.
Automatic Functionality
The sending of RSIP messages at the beginning and the end of the test is configured, and the Enable retransmissions option is selected.
Endpoint2
Scenario The test scenario mirrors the Endpoint1 call flow for the receiving side (called party) and comprises the Wait ReceiveCall, Voice Session, and Wait Recv_EndCall.
Execution Settings The corresponding scenario channel is configured to execute 2 loops during the test sustain time.
Simulated Endpoints
The endpoint phone number is specified using the 160[00000-] sequence generator expression.
SDP No custom SDP settings are configured.
Table 4-144. MGCPGW1 and Endpoint1 Activity Test Settings
Category Settings
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Codecs The G.711 a-law and G.711 u-law codecs are selected.
RTP RTP hardware acceleration is configured.
Audio The Enable audio on this activity option is selected.
SRTP Use of SRTP is not enabled.
Other Settings No scenario variables need to be initialized.
Table 4-145. MGCPCA1 and Endpoint2 Activity Test Settings
Category Settings
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PSTN Sample Test Configurations and Test Scenarios
This section describes the predefined IxLoad Voice Plug-in PSTN sample test configurations (RXFs) and their associated test scenarios.
Used Test Configurations
All sample PSTN tests supplied with IxLoad are based on one of the following two configurations:
• T1/E1 Digital only tests: The PSTN-only test flow is generated by two PSTNDigitalPeer activities that exchange digital T1/E1 signaling and media traffic with each other. Both PSTNDigitalPeer activities are configured on the same PSTN NetTraffic (Figure 4-95).
Figure 4-95. PSTN-Only Topology
Tests from this category can run either in back-to-back mode (these have a B2B string in their name), or against a DUT (VS string in their name), such as a router or voice gateway.
• Mixed SIP/Digital T1/E1 tests: The SIP and Digital T1/E1 test flow is exchanged between VoIPSIPPeer and PSTNDigitalPeer activities that estab-lish between themselves calls with media exchange (Figure 4-96).
Figure 4-96. VoIPSIP and PSTN Topology
PSTN Test Configurations
The following sample PSTN test configuration files are contained in the IxLoad installer:
Note: For a complete description of the supported H.248 Test Library functions, refer to Digital T1/E1 Functions Library on page 3-155.
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PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS
This test, which is based on the configuration shown in Figure 4-95, illustrates the case of calls initiated by the PSTNDigitalPeer1 activity and terminated by the PSTNDigitalPeer2 activity. After successful call setup, bidirectional media is exchanged between the call participants using the Voice Session script function.
The network-level configured settings are T1 CAS, D4 framing and B8SZ line encoding, Immediate variant for both PSTN digital ranges.
The activity-level PSTNDigitalPeer1 and PSTNDigitalPeer2 configured settings are described in Table 4-146.
PSTN_002_B2B_T1_CAS_FGD_D4_B8ZS
This test based on the configuration shown in Figure 4-95 runs in B2B mode.
Table 4-146. PSTNDigitalPeer1 and PSTNDigitalPeer2 Activity Test Settings
Category Settings
PSTNDigitalPeer1
Scenario The associated scenario channel implements a simple call sequence (Make Call, Voice Session, End Call) with bidirectional media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 5 loops during the test sustain time.
Dial Plan The PSTNDigitalPeer source phone numbers are specified using a 150[00000000-] sequence generator expression. The activity is configured to initiate a call to the PSTNDigitalPeer2 activity specified using a symbolic link.
Audio PSTNDigitalPeer media functions are configured to play a specified audio clip for its entire duration.
PSTNDigitalPeer2
Scenario The associated scenario channel implements a simple call sequence for the terminating side (Receive Call, Voice Session, End Call) with media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 5 loops during the test sustain time.
Dial Plan Since this activity only terminates the call, no call destination is configured.
Audio PSTNDigitalPeer media functions are configured to play a specified audio clip for its entire duration.
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This test is similar to the previous PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS one, with the difference that PSTNDigitalPeer activities are configured using the FGD instead of the Immediate variant at network level.
PSTN_003_B2B_E1_CAS_Argentina_G704_HDB3
This test based on the configuration shown in Figure 4-95 runs in B2B mode.
This test is similar to the previous PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS one, with the difference that PSTNDigitalPeer activities are configured using the E1 type, CAS signaling, G704 framing, and HDB3 line encoding at network level.
PSTN_004_B2B_T1_4ESS_D4_B8ZS
This test based on the configuration shown in Figure 4-95 runs in B2B mode.
This test is similar to the previous PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS one, with the difference that PSTNDigitalPeer activities are configured using the T1 type, ISDN PRI signaling, E4 framing, B8ZS line encoding, and 4ESS proto-col at network level.
PSTN_005_B2B_T1_QSIG_D4_B8ZS
This test based on the configuration shown in Figure 4-95 runs in B2B mode.
This test is similar to the previous PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS one, with the difference that PSTNDigitalPeer activities are configured using the T1 type, ISDN PRI signaling, D4 framing, B8ZS line encoding, and QSIG proto-col at network level.
PSTN_006_B2B_T1_5ESS_D4_B8ZS
This test based on the configuration shown in Figure 4-95 runs in B2B mode.
This test is similar to the previous PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS one, with the difference that PSTNDigitalPeer activities are configured using the T1 type, ISDN PRI signaling, D4 framing, B8ZS line encoding, and 5ESS proto-col at network level.
PSTN_007_B2B_E1_ISDN_QSIG_G704_HDB3
This test based on the configuration shown in Figure 4-95 runs in B2B mode.
This test is similar to the previous PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS one, with the difference that PSTNDigitalPeer activities are configured using the E1 type, ISDN PRI signaling, G704 framing, HDB3 line encoding, and QSIG protocol at network level.
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PSTN_008_B2B_E1_ISDN_KHT_G704_HDB3
This test based on the configuration shown in Figure 4-95 runs in B2B mode.
This test is similar to the previous PSTN_001_B2B_T1_CAS_IMM_D4_B8ZS one, with the difference that PSTNDigitalPeer activities are configured using the E1 type, ISDN PRI signaling, G704 framing, HDB3 line encoding, and KHT protocol at network level.
PSTN_009_VS_Cisco_E1_ISDN_vs_E1_ISDN
This test, which is based on the configuration shown in Figure 4-96, illustrates the case of calls initiated by the PSTNDigitalPeer1 activity and terminated by the PSTNDigitalPeer2 activity. After successful call setup, bidirectional media is exchanged between the call participants using the Voice Session script function.
The network-level configured settings are E1 ISDN PRI, G704 framing HDB3 line encoding, QSIG protocol for both PSTN digital ranges.
The PSTNDigitalPeer1 and PSTNDigitalPeer2 configured settings are described in Table 4-147.
Table 4-147. PSTNDigitalPeer1 and PSTNDigitalPeer2 Activity Test Settings
Category Settings
PSTNDigitalPeer1
Scenario The associated scenario channel implements a simple call sequence (Make Call, Voice Session, End Call) with bidirectional media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 5 loops during the test sustain time.
Dial Plan The PSTNDigitalPeer1 source phone numbers are specified using a 150[00000000-] sequence generator expression. The activity is configured to initiate calls to the PSTNDigitalPeer2 activity specified using a symbolic link.
Audio PSTNDigitalPeer1 media functions are configured to play a specified audio clip for its entire duration.
PSTNDigitalPeer2
Scenario The associated scenario channel implements a simple call sequence for the terminating side (Receive Call, Voice Session, End Call) with bidirectional media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 5 loops during the test sustain time.
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PSTN_010_VS_Cisco_SIP_vs_T1
This mixed SIP - Digital T1/E1 test, which is based on the configuration shown in Figure 4-96, illustrates the case of calls initiated by the VoIPSIPPeer1 activity and terminated by the PSTNDigitalPeer1 activity. After successful call setup, bidirectional media is exchanged between the call participants using the Voice Session script function.
The network-level PSTNDigitalPeer1 configured settings are T1 ISDN PRI, 4ESS protocol, ESF framing, B8ZS line encoding for the network range pertain-ing to the activity.
The activity-level VoIPSIPPeer1 and PSTNDigitalPeer1 configured settings are described in Table 4-148.
Dial Plan Since this activity only terminates the call, no call destination is configured. The source phone numbers are specified using the 20[00-] sequence generator expression
Note: This sequence must be present on the DUT the test is run against.
Audio PSTNDigitalPeer1 media functions are configured to play a specified audio clip for its entire duration.
Table 4-148. VoIPSIPPeer1 and PSTNDigitalPeer1 Activity Settings
Category Settings
VoIPSIPPeer1
Scenario The associated scenario channel implements a simple call sequence (Make Call, Voice Session, End Call) with bidirectional media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 1 loop during the test sustain time.
Dial Plan The source phone number is specified using the 7777[00-] sequence generator expression. The IP address of the DUT is configured in the destination Destination IP field, the destination phone numbers are specified using the 9999[00-] expression and the Override phone numbers from destination activity option is selected.
SIP The SIP UAs emulated by this activity are configured to use an outbound SIP proxy server specified using an IP address.
Automatic Retransmissions and timers are not configured.
Codecs The default G.711 a-law and G.711 u-law codecs are selected.
Table 4-147. PSTNDigitalPeer1 and PSTNDigitalPeer2 Activity Test Settings
Category Settings
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PSTN_011_VS_Cisco_T1_CAS_IMM_vs_T1_CAS_IMM
This test based on the configuration shown in Figure 4-95 runs against a DUT.
This test is similar to the previous PSTN_009_VS_Cisco_E1_ISDN_vs_E1_ISDN one, with the difference that PSTNDigital activities are configured using T1, CAS signaling, ESF framing, B8ZS line encoding, and E&M protocol at network level.
PSTN_012_VS_Cisco_T1_ISDN_5ESS_vs_T1_ISDN_5ESS
This test is similar to the previous PSTN_009_VS_Cisco_E1_ISDN_vs_E1_ISDN one, with the difference that PSTNDigital activities are configured using T1 type, ISDN PRI signaling, ESF framing, B8ZS line encoding, and 5ESS protocol at network level.
PSTN_013_VS_Cisco_T1_vs_SIP
This mixed SIP - Digital T1/E1 test, which is based on the configuration shown in Figure 4-96, illustrates the case of calls initiated by the PSTNDigitalPeer1 activity and terminated by the VoIPSIPPeer1 activity. After successful call setup, bidirectional media is exchanged between the call participants using the Voice Session script function.
The network-level PSTNDigitalPeer1 configured settings are T1 ISDN PRI, 4ESS protocol, ESF framing, B8ZS line encoding for the network range pertain-ing to the activity.
RTP No custom RTP settings are configured.
Audio The Enable audio on this activity option is selected.
Other No scenario variables need to be initialized.
PSTNDigitalPeer1
Scenario The associated scenario channel implements a simple call sequence (Receive Call, Voice Session, End Call) with bidirectional media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 5 loop during the test sustain time.
Dial Plan The PSTNDigitalPeer1 source phone numbers are specified using a 9999[00-] sequence generator expression.
Audio PSTNDigitalPeer1 media functions are configured to play a specified audio clip for its entire duration.
Table 4-148. VoIPSIPPeer1 and PSTNDigitalPeer1 Activity Settings
Category Settings
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The activity-level VoIPSIPPeer1 and PSTNDigitalPeer1 configured settings are described in Table 4-149.
Table 4-149. VoIPSIPPeer1 and PSTNDigitalPeer1 Activity Settings
Category Settings
PSTNDigitalPeer1
Scenario The associated scenario channel implements a simple call sequence (Make Call, Voice Session, End Call) with bidirectional media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 5 loop during the test sustain time.
Dial Plan The PSTNDigitalPeer1 source phone numbers are specified using a 9999[00-] sequence generator expression. VoIPSIPPeer1 is configured as destination activity, the destination phone numbers are specified using the 7777[00-] expression, and the Override phone numbers from destination activity option is selected
Audio PSTNDigitalPeer1 media functions are configured to play a specified audio clip for its entire duration.
VoIPSIPPeer1
Scenario The associated scenario channel implements a simple call sequence (Receive Call, Voice Session, End Call) with bidirectional media streaming performed after call setup.
Execution Settings The corresponding scenario channel is configured to execute 1 loop during the test sustain time.
Dial Plan The source phone number is specified using the 7777[00-] sequence generator expression.
SIP The SIP UAs emulated by this activity are configured to use an outbound SIP proxy server specified using an IP address.
Automatic Retransmissions and timers are not configured.
Codecs The default G.711 a-law and G.711 u-law codecs are selected.
RTP No custom RTP settings are configured.
Audio The Enable audio on this activity option is selected.
Other No scenario variables need to be initialized.
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5Chapter 5: Extended Functionality SIP Tests Suite
This chapter covers the following topics:
• SIP Tests Suite Overview on page 5-1.
• Predefined Common Test Procedures on page 5-5
• Test Descriptions on page 5-8
SIP Tests Suite Overview
The Extended Functionality SIP test suite consists of a set of test configurations (IxLoad rxf and tst files) aimed at covering a broad range of SIP device or VoIP network testing needs. The test suite is able to cover variations of the SIP imple-mentations (different types of DUT) by using parameters in the test configura-tions.
Engineers testing SIP implementations use specific test plans focused on the area of interest. Implementing the test plan in the IxLoad configuration – and more generally in any advanced test tool – requires a considerable effort especially for VoIP functionality tests. Having access to a comprehensive set of configuration reduces the time spent in building configurations and allows testers to focus on the DUT, and not on the test tool.
The customers can use the test suite by selecting the subset of tests that match their specific test plan, changing the specific, DUT-related parameters, and exe-cute tests using the IxLoad Voice Plug-in module. Although in some cases a customization of the call flow and SIP message parameters would still be needed, this will be much easier that the approach whereby the configurations have to be created based on the supplied set of samples.
General Test Features
The supplied test configurations share the following specifications:
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General Settings
• The configurations (rxf) are provided for Acceleron-XP load module cards, having one chassis port allocated to each VoIPSIP activity.
• For all configurations, a single test scenario per configuration is used, whereby the name of the test scenario is the same as that of the configuration, but with a different extension.
Network Settings
• The values of the test parameters – IP addresses for NetTraffics, the IP address of the SIP Proxy/Registrar server, the dial plans, the domain names – are consistent across the configurations. The used network addresses values are 20.1.1.1/16 for the first VoIPSIP activity, 20.1.50.1 for the second, and so on, 20.1.254.254 for the SIP Proxy/Registrar IP address, and ixload-test.com for domain name.
Generally, tests configured with the Use consecutive values (per port) IP address allocation scheme configured in the VoIPSIP activity’s Execution page have 8000 IP addresses defined (starting from 20.1.1.1/16), while those using the Use the same value (per port) IP address allocation scheme have 12 IP addresses defined (starting from 20.1.1.1/16).
In order to prevent the flooding of testbed switches, the sending of Gratuitous ARP messages is disabled, and ARP requests are made at test execution time for the emulated SIP UAs only.
Activity Settings
• The test configurations with media exchange (RTP) have audio (voice) full duplex support using the G.711 codec; with hardware acceleration enabled. For tests with the LPS objective type that use media script functions, these are configured to play media for the duration of the TalkTime call parameter.
• Generally, tests configured using an AC or LPS objective have the Use con-secutive values (per port) IP address allocation scheme configured in the VoIPSIP activity’s Execution page.
Figure 5-1. IP Address Allocation Settings for LPS Tests
• Test configured with a Channels objective use multiple SIP UAs that share the same IP address (the Channel mapping for SIP UAs parameter in the VoIPSIP activity’s Execution page is set to Use same value (per port) value). The media address is also shared by all UAs (the Channel mapping for media parameter is set to Use same value (per port) value), while the media port uses consecutive values.
These tests are identified by the MCH suffix appended to the test name.
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Figure 5-2. IP Address Allocation Settings for MCH Tests
• For tests that have an AC or LPS objective type, media functions used in the test scenario are configured for playing audio/video media for the duration of the TalkTime call parameter (Play for clip duration or TalkTime option in the Audio page).
For tests that use a Channels objective type, media functions are configured for playing audio/video media for a specified duration of time (seconds).
Scenario Settings
• All call flows include a SIP REGISTER message that is executed only once, on the first iteration of the test. Session timers for the Register operation are enabled at activity level in the VoIPSIP activity’s Automatic page (the Expires header is configured to 3600 seconds, re-registrations are handled automatically every 3500 seconds).
• Most of the call flows are created so as to support the SIP Route/Record-Route mechanism. The use of Route headers on requests (other than the sce-nario’s first INVITE) and Record-Route headers on responses inside Make Call, Receive Call, End Call, or other procedures provides support for test-ing Proxy servers that choose to stay in the message path between call partic-ipants.
Whenever a procedure uses SIP Route for requests or Record-Route headers for responses, the Route or RecordRoute suffixes are appended to the proce-dure name.
• All call flows were created using a SIP Allow header that advertises the INVITE, ACK, BYE, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, REFER, MESSAGE methods.
• For most request messages, the SIP User-Agent header is present and the User-Agent value is defined in the SIP REGISTER message using the “IxLoad-client”+$UnitCh+”/v5.10” expression.
• All call flows were created so as to handle a large number of response mes-sages from the 1xx, 2xx, 4xx, 5xx, 6xx classes, or the lack of response from test devices.
Test Objective Settings
• Configurations are provided for capacity and performance testing, the config-ured test objectives being Channels, Active Callers (AC), or Loops Initiated per Second (LPS) respectively.
Generally, tests configured using an LPS objective type have a number of channels specified in the Custom Parameters tab of the Timeline and Objectives page (see figure below). The resulting TalkTime call parameter is
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used for configuring one occurrence of media functions (Talk, Listen, or Voice Session) executed by the script scenario. For additional occurrences of media functions, the play time is specified in other modes than by using the TalkTime parameter.
Figure 5-3. LPS - Custom Parameters
The Estimated Overhead Time parameter configured in the Custom Parame-ters tab takes into account delays introduced by the additional media func-tions, as well as Sleep functions or other functions that generate delays, used in a particular test scenario.
• Generally tests are configured using a RampUp time of 2 minutes (RampUp value of 25 users/s), a SustainTime of 1 hour, and a RampDown time of 5 minutes (RampDown value of 50 users/s).
Test Categories The supplied test configurations fall into following categories:
• Registration
• Basic Calls
• Advanced SIP Features
For each test, the test configuration is described and information is provided on how to start from the existing configuration and adapt the test to your own needs.
Customizable Test Parameters
Although test configurations contained in the SIP Extended Functionality test suite are already configured, the following test parameters can be easily modified in order to adapt a test to your own testbed:
• IP address: The IP addresses are configured in a VoIPSIP activity’s underly-ing Network page. Initially, test configurations use IP addresses starting with 20.1.1.1/16 for the first VoIPSIP activity, with 20.1.50.1/16 for the second VoIPSIP activity, and so on.
• Phone numbers: The source phone numbers are configured in a VoIPSIP activity’s Dial Plan page. Initially, test configurations use the 160[00000000-] and 170[00000000-] sequence generating expressions for source and destina-tion phone numbers.
• Proxy server IP address: For tests that use an external server, its IP address is configured in the VoIPSIP activity’s SIP page.
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• Domain name: Emulated UAs have the ixload-test domain configured in the VoIPSIP activity’s SIP page.
• Chassis port: VoIPSIP activities are configured in the Port Assignments page using a single port of an Acceleron load module board.
Predefined Common Test Procedures
Tests use some common registration and call procedures – Register and Make call – described in this section.
Register The Register procedure (Figure 5-4) performs registration on the first loop.
Figure 5-4. Register Procedure
The initial Variable Test function tests if the current iteration is the first one, so as to execute registration only once, at the beginning of the test. The following Clear Statistics function resets the reg_sent_err and reg_recv_err variables to an empty (““) value.
The actual registration is done by the Register procedure illustrated in Figure 5-5:
Figure 5-5. Register Procedure
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A SIP REGISTER message is sent using the Send Register function, then the count variable (which represents the occurrence of received 401 response mes-sage) is initialized to the ‘1’ value.
The following Wait... function processes a large number of responses:
• For 1xx and 491 messages, no action is taken and the Wait... function loops.
• For 200 Ok messages, a Sleep function is executed and the Register proce-dure exits.
• For 401 messages, a Count 401 procedure verifies if the received 401 mes-sage is the first one; if this is the case, the REGISTER request is re-sent with authentication information and the script execution jumps to the Wait... func-tion. Alternatively, in case this is a second 401 message received, which rep-resents a permanent registration failure, the script execution starts anew from the initial Send Register function.
• For 407 messages, a Count 407 procedure verifies if the received 407 mes-sage is the first one; if this is the case, the REGISTER request is re-sent with authentication information and the script execution jumps to the Wait... func-tion.
• For 423 messages, first an Extract Variable function is used for extracting the new_exp variable, then the REGISTER request is re-sent with an Expires header configured to this value.
• For 4xx messages other than 401, 407, 423, and 491, a message is logged using the Log Message function, then the reg_sent_err and reg_recv_err variables (containing the sent and received error messages) are extracted using two Extract Variable functions inside the Save_error_messages pro-cedure. The extracted strings are eventually written to logs using the Dump Variable function.
• For 5xx and 6xx error messages, Log Message functions are triggered that log specific messages to the chassis port and which are displayed the IxLoad Event Viewer window.
When the Register procedure encounters an error and exits on its Error output, a message is logged and the registration operation resumes, following an idle period enforced by a Sleep function.
Make Call The Make Call procedure (Figure 5-6) establishes a call.
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Figure 5-6. Make Call Procedure
The procedure contains an initial Clear_Variables function that resets the proxy_auth string variable to an empty value. The call is initiated using an Invite function, followed by a Wait function that processes a large number of response messages:
• For 1xx messages, no action is taken and the Wait... function loops.
• For 200 Ok messages, an ACK (call connected) procedure is executed that signals a successful call establishment.
• For 401 messages, an Authorization procedure is executed that first sends an ACK message and then re-sends the INVITE message with authentication information.
• For 407 messages, a Proxy-Authorization procedure is executed that sends an ACK message and then re-sends the INVITE message with authentication information.
• For 491 messages, a message is logged using the Log Message function and then execution resumes starting from the Wait... function.
• For 4xx (other than the above), 5xx, and 6xx error messages, an ACK (error) procedure is triggered that logs an error message and then extracts the call_sent_err and call_rec_err messages using Extract Variable functions. Finally these error messages are dumped to the port (and to the Event Viewer pane) using a Dump Variables script function.
• For BYE messages, a 200 Ok procedure is executed and then the procedure exits on the Error output.
UnRegister The UnRegister procedure (Figure 5-7) performs de-registration while the test execution is on the ramp-down portion.
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Figure 5-7. UnRegister Procedure
The procedure uses a Variable Test function to test if the execution is on the ramp-down portion ($rampdown variable is equal to 1), then executes the de-reg-istration of SIP UAs.
Test Descriptions
For each test, the test configuration is described and information is provided on how to start from the existing configuration and adapt the test to your own needs.
Registration The script in this category illustrates the registration of IxLoad-emulated SIP UAs with a Registrar server.
Register
This test illustrates the case of SIP UAs that register with a Registrar server.
Basic Calls The scripts in this category illustrate the case of IxLoad-emulated SIP UAs that perform basic calls. Prior to initiating a call, UAs perform registration with a real Registrar using the Register Complete procedure.
Basic_Call_Complete_Audio_LPS
This test illustrates UAs that establish calls to other UAs. After call establish-ment, endpoints exchange media using the Voice Session script function.
At activity level no special settings are configured in addition to those described in General Test Features on page 5-1.
At scenario level, registration is performed using the Register procedure (see Register on page 5-5), while re-registration is configured using the Enable ses-sion timers settings of the Automatic page. Call setup is done using the Make Call procedure (see Make Call on page 5-6). Following the media exchange, end-points finally de-register using an Unregister procedure (see UnRegister on page 5-7).
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
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The configured test objective is 200 Loops Initiated per Second (LPS).
Basic_Call_Complete_Audio_MCH
This test is similar with the Basic_Call_Complete_Audio_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signal-ing and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
Basic_Call_Complete_Multimedia_AC
This test is similar to Basic_Call_Complete_Audio_LPS, with the difference that endpoints involved in the call exchange both audio and video media. The config-ured objective is 100 Active Callers.
Basic_Call_Complete_Multimedia_MCH
This test is similar with the Basic_Call_Complete_Multimedia_AC test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
Basic_Call_Busy_LPS
This test illustrates SIP UAs that attempt o establish calls to other UAs, with the result that the connection is not established and a Busy 486 message is received instead.
At activity level, Caller and Callee register each with a Registrar server specified by an IP address and use an outbound Proxy server (SIP configuration page). Both parties have audio capability enabled (Audio configuration page).
At scenario level, the Caller attempts to establish a call with Callee using a Make Call procedure (see Make Call on page 5-6), while the Callee responds by send-ing a Busy 486 message within a Receive Call-Busy Here procedure.
The configured test objective is 100 Loops Initiated per Second (LPS).
Basic_Call_Busy_MCH
This test is similar with the Basic_Call_Busy_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
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Basic_Call_Complete_Cancel_LPS
This test illustrates the case of SIP UAs that attempt o establish calls to other UAs, whereby the remote party does not answer the incoming call; the Caller eventually cancels the call request.
At activity level, Caller and Callee register each with a Registrar server specified by an IP address and use an outbound Proxy server (SIP configuration page). Both parties have the audio capability enabled (Audio configuration page).
At scenario level, the Caller attempts to establish a call with Callee using a Make Call procedure. Since the remote party does not answer the incoming call, Caller eventually sends a SIP CANCEL message (using a CANCEL Call procedure) to cancel the request.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parameters tab.
Basic_Call_Complete_Cancel_MCH
This test is similar with the Basic_Call_Complete_Cancel_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both sig-naling and media, i.e. the Channel mapping rules for SIP UAs and Channel map-ping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 4000 Channels.
Advanced SIP Features
The scripts in this category illustrate the case of IxLoad-emulated SIP UAs that perform calls with other emulated UAs, whereby advanced SIP features, such as Call Hold, Call Transfer, Call Park, or 3-Way Calls, are used after call establish-ment.
Call_Hold_LPS
This test implements the call flow shown in Figure 5-8, whereby the Caller and the Callee are emulated by Ixload VoIPSIP activities, while the Proxy is a real device (DUT).
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Figure 5-8. Call Hold
At activity level, Caller and Callee register each with a Registrar server specified by an IP address and use an outbound Proxy server (SIP configuration page). Both parties have the audio capability enabled (Audio configuration page).
At test scenario level, UAs establish calls to other UAs using the Make Call pro-cedure (see Make Call on page 5-6) and then exchange audio media over the call using the Voice Session function. The remote party then puts the call on hold (SIP Hold-Initiate procedure) for a duration configured by the Sleep function and eventually takes the call off the hold state (SIP Unhold-Initiate procedure). Finally endpoints exchange media again over the re-established call using the Voice Session function.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 3000 channels configured in the Custom Parameters tab.
Call_Hold_MCH
This test is similar with the Call_Hold_LPS test, except for fact that multiple SIP UAs are configured to share the same IP address, i.e. the IP address for SIP UAs parameter in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
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The configured test objective is 4000 Channels.
Consultation_Hold_LPS
This test implements the call flow shown in Figure 5-9, whereby the Caller, Callee1, and Callee2 are emulated by Ixload VoIPSIP activities, while the Proxy is a real device (DUT).
Figure 5-9. Consultation Hold
At activity level, Caller, Callee1, and Callee2 register each with a Registrar server specified by an IP address and use an outbound Proxy server (SIP configu-ration page). All three parties have audio capability enabled (Audio configuration page).
At test scenario level, UAs establish calls to other UAs using the Make Call pro-cedure and then exchange audio media over the call (Voice Session function). Callee1 then puts the call with Caller on hold (SIP Hold-Initiate procedure) and establishes a call to another destination (Callee2) for a period of time configured using the Sleep function. The call between destination (1) and (2) is terminated
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by Callee1 using the SIP EndCall Initiate - Route procedure (BYE message contains a Route header), after which Callee takes the initial call off hold and exchanges again media with Caller using the Voice Session function.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 3000 channels configured in the Custom Parameters tab.
Consultation_Hold_MCH
This test is similar with the Consultation_Hold_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Music_on_Hold_LPS
This test implements the call flow shown in Figure 5-10, whereby the Caller, the Callee, and the Music Server are emulated by Ixload VoIPSIP activities.
Figure 5-10. Music On Hold
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At activity level, Caller and Callee1 register each with a Registrar server speci-fied by an IP address and use an outbound Proxy server (SIP configuration page). All three parties have audio capability enabled (Audio configuration page). As a note, the Music server is configured using the Use same value (per port) setting for both SIP and RTP IP address allocation (Execution page).
At test scenario level, Caller establishes a call to Callee using the Make Call pro-cedure and exchanges audio media over the call (Voice Session function). The Callee then sends an INVITE to the Music Server without an SDP definition and puts the call with Caller on hold (SIP Hold-Initiate procedure). The Music server streams media to Caller (using the Talk function), before Caller finally terminates the call using a SIP EndCall Initiate procedure.
Eventually the call between Callee and Caller is resumed by Callee who sends a re-INVITE to Caller using a SIP Unhold - Initiate procedure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parameters tab.
Music_on_Hold_MCH
This test is similar with the Music_on_Hold_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Transfer_Unattended_LPS
This test implements the call flow shown in Figure 5-11, whereby the Caller, Callee1, and Callee2 are emulated by Ixload VoIPSIP activities.
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Figure 5-11. Unattended Transfer
At activity level Caller, Callee1, and Callee2 register each with a Registrar server specified by an IP address and use an outbound Proxy server (SIP configuration page). All three parties have audio capability enabled in the Audio configuration page.
At test scenario level, Caller establishes a call to Callee1 using the Make Call procedure and then exchange audio media over the call (Voice Session function). Callee1 then REFERs the Caller to Callee2 (Initiate Transfer procedure), with whom the Caller establishes a call and exchanges audio media. Finally the Caller uses the Confirm Transfer procedure to notify the transfer completion to Callee1. The call is terminated by Callee2 using a SIP EndCall Initiate proce-dure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 50 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parame-ters tab.
Transfer_Unattended_MCH
This test is similar with the Transfer_Unattended_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping
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rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Transfer_Attended_LPS
This test implements the call flow shown in Figure 5-12, whereby the Caller, Callee1, and Callee2 are emulated by Ixload VoIPSIP activities. The test is simi-lar to the previous one, except that a consultation with Callee2 occurs before transferring the call.
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Figure 5-12. Attended Transfer
At activity level, Caller, Callee1 and Callee2 register each with a Registrar server specified by an IP address and use an outbound Proxy server (SIP configuration page). All three parties have audio capability enabled in the Audio configuration page.
At test scenario level, Caller establishes a call with Callee1 using the Make Call procedure and then exchange audio media over the call (Voice Session function). Callee1 then puts the call with Caller on hold (SIP Hold-Initiate procedure) and establishes a consultation session with Callee2. After performing media exchange, Callee1 puts the call with Callee2 on hold and REFERs Caller to Callee2 (Initiate Transfer procedure). Caller replaces Callee1 (Make Call pro-cedure containing an INVITE message with a Replaces header) and establishes a session with Callee2. This call is terminated by Callee2 using a SIP EndCall Initiate procedure.
Finally Callee1 terminates the Call with Caller using a similar SIP EndCall Initiate procedure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 50 Loops Initiated per Second (LPS), which is to be achieved using a number of 1500 channels configured in the Custom Parame-ters tab.
Transfer_Attended_MCH
This test is similar with the Transfer_Attended_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Transfer_Instant_Messaging_LPS
This test implements the call flow shown in Figure 5-13, whereby the Caller, Callee1, and Callee2 are emulated by Ixload VoIPSIP activities.
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Figure 5-13. Transfer via Instant Messaging
At activity level, Caller, Callee1 and Callee2 register each with a Registrar server specified by an IP address and use an outbound Proxy server (SIP configuration page). All three parties have audio capability enabled (Audio configuration page).
At test scenario level, Caller establishes a call with Callee1 using the Make Call procedure and the parties exchange audio media over the call using the Voice Session function. Dialog information is then provided by Callee1 to Callee2 using a Send Instant Message procedure (containing a MESSAGE command). Callee2 sends Caller an INVITE message (using a Make Call procedure) that contains a Replaces header and the two parties establish a session with audio media exchange. The call is eventually terminated by Callee2 using a SIP End-Call Initiate procedure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parameters tab.
Transfer_Instant_Messaging_MCH
This test is similar with the Transfer_Instant_Messaging_LPS, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
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Call_Forwarding_Unconditional_LPS
This test implements the call flow shown in Figure 5-14, whereby the Caller and the Gateway are emulated by Ixload VoIPSIP activities, while the Proxy is a real device (DUT).
Figure 5-14. Unconditional Call Forwarding
At activity level, the Caller registers with the Registrar server specified by its IP address and uses an outbound Proxy server (SIP configuration page). Both the Caller and the Gateway have audio capability enabled (Audio configuration page).
At scenario level, the Caller tries to place a call to a PSTN number, while the Proxy server routes the call to a Gateway. On the established call, media is exchanged using the Voice Session function. Eventually the Caller terminates the call sending a BYE message (within the SIP EndCall Initiate procedure) that contains a Route header.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parameters tab.
Call_Forwarding_Unconditional_MCH
This test is similar with the Call_Forwarding_Unconditional_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
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Call_Forwarding_on_Busy_LPS
This test implements the call flow shown in Figure 5-15, whereby the Caller, Callee(id1), and Callee(id2) are emulated by Ixload VoIPSIP activities, while the Proxy is a real device (DUT).
Figure 5-15. Call Forwarding on Busy
At activity level, Caller and Callee register each with a Registrar server specified by its IP address (SIP configuration page). Both Callee identities have the same phone number configured (Dial Plan page), and both the Caller and the Callee have audio capability enabled (Audio configuration page).
At scenario level, the Caller attempts to place a call to Callee, with the Proxy routing the call first to Callee (id1),who replies with a ‘486 Busy here’ response, and then to Callee (id2). The call is accepted by Callee (id2) using a SIP Receive Call procedure that contains ‘180 Ringing’ and ‘200 OK’ messages. After exchanging media over the established call using the Voice Session function, the Caller terminates the call sending a BYE message containing a Route header (SIP EndCall Initiate - Route procedure).
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parameters tab.
Call_Forwarding_on_Busy_MCH
This test is similar with the Call_Forwarding_on_Busy_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signal-ing and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping
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rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Call_Forwarding_on_No_Answer_LPS
This test implements the call flow shown in Figure 5-16, whereby the Caller, Callee(id1) and Callee(id2) are emulated by Ixload VoIPSIP activities, while the Proxy is a real device (DUT).
The test is similar to the previous Call_Forwarding_on_Busy_LPS test, except that the call is forwarded by the Proxy server on a no answer timeout condition instead of a busy condition.
Figure 5-16. Call Forwarding on No Answer
Call_Forwarding_on_No_Answer_MCH
This test is similar with the Call_Forwarding_on_No_Answer_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Three_Way_Conference-Third_Party_Added_LPS
This test implements the call flow shown in Figure 5-17, whereby the Caller, Callee1, and Callee2 are emulated by Ixload VoIPSIP activities.
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Figure 5-17. Three Way Conference - Third Party Added
At activity level, Caller, Callee1, and Callee2 register each with a Registrar server specified by its IP address (SIP configuration page). All VoIPSIP activi-ties have audio capability enabled (Audio configuration page).
At scenario level, the Caller places a call to Callee1 and the party exchange audio media. Callee1 then uses a Make Call procedure to re-invite Caller, then uses another Make Call procedure for inviting Callee2.
After exchanging media over the established call using the Voice Session func-tion, the Caller terminates the call sending a BYE message within SIP EndCall Initiate - Route procedure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 1000 channels configured in the Custom Parameters tab.
Three_Way_Conference-Third_Party_Added_MCH
This test is similar with the Three_Way_Conference-Third_Party_Joins_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Exe-cution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
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Three_Way_Conference-Third_Party_Joins_LPS
This test implements the call flow shown in Figure 5-18, whereby the Caller, Callee1, and Callee2 are emulated by Ixload VoIPSIP activities.
Figure 5-18. Three Way Conference - Third Party Joins
At activity level, Caller, Callee1, and Callee2 register each with a Registrar server specified by its IP address (SIP configuration page). All VoIPSIP activi-ties have audio capability enabled (Audio configuration page).
At scenario level, the Caller places a call to Callee1 and the parties exchange audio media. Callee2 joins the conference using a Make Call procedure that con-taining an INVITE message with a Join header. Callee1 also re-INVITEs Caller to the focus mode using another Make Call procedure.
After exchanging media over the established call using the Voice Session func-tion, the Caller terminates the call sending a BYE message within a SIP EndCall Initiate - Route procedure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 1000 channels configured in the Custom Parameters tab.
Three_Way_Conference-Third_Party_Joins_MCH
This test is similar with the Three_Way_Conference-Third_Party_Joins_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs
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and Channel mapping rules for media parameters in the VoIPSIP activity’s Exe-cution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Single_Line_Extension_LPS
This test implements the call flow shown in Figure 5-19, whereby the Caller, Callee(id1), Callee(id2), and Callee(id3) are emulated by Ixload VoIPSIP activi-ties, and the Forking Proxy is a real device (DUT).
Figure 5-19. Single Line Extension
At activity level, all Callee identities have the same phone number configured.
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At scenario level, Caller tries to place a call to Callee, while the Proxy routes the call in turn to all known identities (id1, id2, id3, and id4), until the call is estab-lished successfully. Callee(id3) sends a NOTIFY message containing the dialog information to Callee’s Address-of-Record, and Callee(id2) attempts to join the call using an INVITE message with a Join header. The request is accepted by Callee(id3) who perform the media mixing.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 1000 channels configured in the Custom Parameters tab.
Single_Line_Extension_MCH
This test is similar with the Single_Line_Extension_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Find-Me_LPS
This test implements the call flow shown in Figure 5-20, whereby the Caller, Callee(id1), Callee(id2), and Callee(id3) are emulated by Ixload VoIPSIP activi-ties, while the Proxy server is a real device.
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Figure 5-20. Find Me Call
At activity level all Callee identities are configured using the same phone num-ber.
At scenario level, Caller tries to place a call to Callee using a Make Call proce-dure, while the Proxy routes the call in turn to all known Callee identities (id1, id2, id3, and id4), until the call is established successfully with Callee(id4). After exchanging media using the Voice Session function, Callee(id4) terminates the call by sending a BYE message within a SIP EndCall Initiate - Route proce-dure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 1000 channels configured in the Custom Parameters tab.
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Find-Me_MCH
This test is similar with the Find-Me_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Call_Park_LPS
This test implements the call flow shown in Figure 5-21, whereby the Caller, Callee1, and Callee2 are emulated by Ixload VoIPSIP activities, and the Park server is a real device (DUT).
Figure 5-21. Call Park
At activity level no special settings are configured other than those described in General Test Features on page 5-1.
At scenario level, Caller and Callee1 establish a call with audio media exchange, then Callee1 parks the call by sending a REFER (Replaces=<call-id>) message to the Park server. The Park server NOTIFYs Callee1 that he is trying to com-plete the operation and sends to the Caller an INVITE (Replaces:<call-id>) mes-
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sage, also containing an SDP definition. Caller accepts the request and music is streamed from the server over the established session using a Talk function.
Finally, Callee2 wishes to retrieve the call and sends an INVITE with a Replaces header to Caller (Make Call procedure). After being accepted by Caller, the Park server is replaced by Callee2 in the call.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 50 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parame-ters tab.
Call_Park_MCH
This test is similar with the Call_Park_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Call_Pickup_LPS
This test implements the call flow shown in Figure 5-22, whereby the Caller, Callee1, and Callee2 are emulated by Ixload VoIPSIP activities.
Figure 5-22. Call Pickup
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At activity level no special settings are configured other than those described in General Test Features on page 5-1.
At scenario level, using a Make Call procedure Caller places a call to Callee1, who is configured in the same call group as Callee2. Callee2 sends a SUBSCRIBE (Event:dialog) message in order to be able to pick up calls addressed to Callee1. When Callee1 is then called, a NOTIFY message with the dialog information (dialog-info field) is sent to Callee2, who replaces Callee1 in the call with Caller.
The call between Caller and Callee2 is eventually terminated by Caller who sends a BYE message within a SIP EndCall Initiate - Route procedure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parameters tab.
Call_Pickup_MCH
This test is similar with the Call_Pickup_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Automatic_Redial_LPS
This test implements the call flow shown in Figure 5-23, whereby the Caller and the Callee are emulated by Ixload VoIPSIP activities.
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Figure 5-23. Automatic Redial
At activity level no special settings are configured other than those described in General Test Features on page 5-1.
At scenario level, after having made an initial call that results in a busy condition, Caller sends a SUBSCRIBE (Event=dialog) message and then waits for a NOTIFY message using the Wait for Availability Information procedure. Fol-lowing the receiving of the notification, Caller establishes the call using a Make Call procedure. The established call is eventually terminated by Caller who sends a BYE message within a SIP EndCall Initiate - Route procedure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 2000 channels configured in the Custom Parameters tab.
Automatic_Redial_MCH
This test is similar with the Automatic_Redial_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
Click_to_Dial_LPS
This test implements the call flow shown in Figure 5-24, whereby the Caller, the Caller’s PC, and the Callee are emulated by Ixload VoIPSIP activities.
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Figure 5-24. Click to Dial
At activity level, the Caller has no destination phone number configured, with this information being conveyed to him by a REFER message.
At scenario level, a Refer To procedure is used for referring the Callee to Caller. Caller then uses a Make Call procedure to establish the call and exchanges audio media using a Voice Call function. The established call is eventually terminated by Caller who sends a BYE message within a SIP EndCall Initiate - Route pro-cedure.
The test scenario provides support for the SIP Route/Record-Route mechanism as described in Scenario Settings on page 5-3.
The configured test objective is 100 Loops Initiated per Second (LPS), which is to be achieved using a number of 1000 channels configured in the Custom Parameters tab.
Click_to_Dial_MCH
This test is similar with the Click_to_Dial_LPS test, except for the fact that SIP UAs are configured as sharing the same IP address for both signaling and media, i.e. the Channel mapping rules for SIP UAs and Channel mapping rules for media parameters in the VoIPSIP activity’s Execution page is set to Use the same value (per port) value.
The configured test objective is 8000 Channels.
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AAppendix A: Creating a SIP Message from Template
The Create from Template window helps you create correct SIP messages by providing templates for SIP message parameters.
When creating a new SIP message using the Create from Template window, you can take either of the approaches below:
• Start from a generic message and select the headers to include in the SIP mes-sage
• Load and subsequently modify a predefined message template.
Table A-1 to Table A-3 describe the main parameters and options you can set in the Structured Message window. For more details about these parameters, please refer to:
• RFC 3261 – Session Initiation Protocol
• RFC 3265—Session Initiation Protocol—Specific Event Notification
• RFC 3515 – The Session Initiation Protocol Refer Method
• RFC 3262 – Reliability of Provisional Responses in SIP
• RFC 2976 – The SIP INFO Method
• RFC 3311 – The SIP Update Method
• RFC 3455 – Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP)
• RFC 3325 – Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks
• draft - ietf - sip - replaces - 04
• draft - ietf - referred by -05
• draft - ietf - sipping - cc - transfer - 02
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SIP Message Elements
Depending on the message type, SIP messages comprise the following elements:
• A Request line (request) or a Status Line (response)
• A variable number of message headers
• Message body
While the message headers are common to both request and response messages, first line parameters are different, depending on whether the message is a SIP request or a SIP response.
Table A-1 on page A-3, Table A-2 on page A-3, and Table A-3 on page A-5 list the specific request and response parameters, as well as the common parameters.
• Specific Request Parameters
• Specific Response Parameters
• Common Parameters
Specific Request Parameters
SIP Request parameters are the request line and the message headers parameters.
When clicking the Create From Template button for a request message, the specific request parameters listed in Table A-1 on page A-3 are available.
Note: The Create from Template window enables you only to build the message first line and the message headers part, it has no influence on defining the message body.
NOTE: The Parameters page contains information about the structure and content of the message body, but in order to determine how the message looks like, you must visit the Behavior and Flow Manager pages.
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Specific Response Parameters
Table A-1. Specific Request - Parameters
Option Description
Current Template
The message template name that is currently loaded
Load Template
Opens the Load SIP Message Template dialog and enables you to select the SIP message template to load. The available templates are:
• <Generic>
• INVITE
• ACK
• OPTIONS
• BYE
• CANCEL
• REGISTER
• NOTIFY
• SUBSCRIBE
• REFER
• MESSAGE
• PRACK
• INFO
• UPDATE
Parameter (list)
The list with available parameters and the appropriate settings. The Request-Line comprises the following main request parameters:
• Method
• Request-URI
• SIP-Version
Table A-2. Specific Response - Parameters
Option Description
Current Template
The SIP status template name that is currently loaded. The status templates are defined as combinations of SIP message templates and status codes.
SIP messages templates:
• <Generic>
• INVITE
• ACK
• OPTIONS
• BYE
• CANCEL
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SIP Message ElementsA
Common Parameters
In addition to the first line of the message—a Request line for requests and a Sta-tus line for responses—SIP messages comprise message headers, which are com-mon to both requests and responses. Table A-3 on page A-5 includes a description of the common parameters.
• REGISTER
• NOTIFY
• SUBSCRIBE
• REFER
• MESSAGE
• PRACK
• INFO
• UPDATE
Load Template Opens the Load SIP Template dialog.
Available message templates and response codes are mentioned above.
Parameters (list)
The list of available parameters and the appropriate settings. The main Status line response parameters are:
• SIP-Version
• Status-Code
• Reason-Phrase
Table A-2. Specific Response - Parameters (Continued)
Option Description
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Table A-3. Common Parameters Description
Option Description
Parameters (list)
Available parameters and appropriate settings:
Message Headers:
Accept, Accept-Encoding, Accept-Language, Alert Info, Allow, Authentication–Info, Authorization, Call-ID, Call-Info, Contact, Content-Disposition, Content-Encoding, Content-Language, Content-Length, Content-Type, Cseq, Date, Error-Info, Event, Expires, From, In-Reply-To, Max-Forwards, Min-Expires, MIME Version, Organization, P-Associated-URI, P-Called-Party-ID, P-Visited-Network-ID, P-Access-Network-Info, P-Charging-Function-Addresses, P-Charging-Vector, P-Asserted-Identity, P-Preferred-Identity, Priority, Proxy-Authenticate, Proxy-Authorization, Proxy-Require, Record-Route, Referred-By, Refer-To, Replaces Reply-To, Require, Retry-After, Route, Server, Subject, Subscription-State, Supported, Timestamp, To, Unsupported, User-Agent, Via, Warning, WWW-Authenticate, extension-header.
NOTE: RSeq and Rack message headers are used only by PRACK and 1xx responses.
NOTE: The following private headers (P-headers) define specific extensions to SIP required for IMS architecture (3GPP) testing:
• P-Associated-URI (RFC 3455)
• P-Called-Party-ID (RFC 3455)
• P-Visited-Network-ID (RFC 3455)
• P-Access-Network-Info (RFC 3455)
• P-Charging-Function-Addresses (RFC 3455)
• P-Charging-Vector (RFC 3455)
• P-Asserted-Identity (RFC 3325)
• P-Preferred-Identity (RFC 3325)
By default, each message header has an index (for example, (#01)) included in its name string. This is because the user can duplicate an existing message header, in which case the duplicate has the number incremented (for example, (#02)).
Color Coding The following color codes are used to show different headers:
DuplicateDuplicates the selected message header.
DeleteDeletes the selected message header. This option is available only if the message has at least one copy.
Move UPMoves the selected message header up.
Move Down
Moves the selected message header down.
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SIP Message ElementsA
NOTE: Moving Up/Down the headers establishes the order in which they are matched by the Wait functions with the defined templates.
Sort Headers
Changes the order in which the headers display. The available options are:
- By Importance – Displays first the important headers.
- By name – Displays headers in alphabetical order.
Show Headers
Chooses the headers to display. There are several types of headers—each displayed in a different color (see color codes). Choose Show Minimal or any combination of the following:
• Mandatory
• TCP Mandatory
• Advisable
• Required if SDP
• Conditional
• Optional
• Invalid
PreviewShows the SIP message that is generated using the current configuration.
Create Saves the current message configuration in a file. If saved for the first time, a window prompts you to enter the file name. By default, the name includes the time and date at that moment.
Load Loads an existing message configuration.
Save Saves the current message configuration in a different file.
OK Creates the ISP message and closes the Structured Message window.
Cancel Discards the changes and closes the window.
Table A-3. Common Parameters Description (Continued)
Option Description
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BAppendix B: The Expression Evaluator Syntax
The SetVar and TestVar script functions of the Flow test library enable you to define complex expressions – based on numerals, variables, operators, functions, and variables – that are evaluated at test execution time and activate different function outputs depending on the result of the evaluations.
This appendix described the entities supported by the Expression Evaluator that is part of these script functions.
Data Types
Supported data types are listed in Table B-1.
Table B-1. Data Types
Type Examples Notes
Number 12
33.4523
Numbers can be integers or floating-point numbers.
String “Welcome to the Voice Mail System”
Strings are always written between inverted commas.
Date You can create data objects using the str2date(“15-oct-01 14:32”) function or you get date objects by reading them from a database record:
$Recordset1.DateTime
To obtain the current date, use the GetCurrentDate() function. To get the components of a date (day, month, hours, and so on) you can use the dtGet___ functions.
For further information, please refer to the Functions section.
NOTE: The date string format depends on the regional settings.
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OperatorsB
Operators
Supported operators—arithmetic, relational, logical, and format operators—are listed in Table B-2 through Table B-5.
Arithmetic Operators Table B-2. Arithmetic Operators
Operator Description Usage Examples Notes
+ Addition • Num-ber+Num-ber
• String + String
• String + Number
• Date + Number
• Date + String
20 + 5 =25
“Channel: “ + $MapPos is assessed as “Channel: 12”
($MapPos is a system variable and its value is equal to the logical channel index).
For numbers, it performs arithmetic addition. For strings, it performs concatenation.
Any operation involving strings gives a new string (concatenation).
The number added to a date represents the number of days (or fraction of days —3.5 days is 3 days and 12 hours—). The result is a new date.
- Subtraction Number–Number
Date – Date
Date – Number
20 - 5 =15
str2date (“2-nov-01”) – 1 is assessed as 1 nov 2001 (date)
str2date(“2-nov-01”) – str2date (“1-nov-01”) = 1
Arithmetic substraction.
The difference between two dates is a number of days.
“Date - number” gives a new date.
* Multiplication
Number * Number
20 * 5 = 100 Arithmetic multiplication
/ Division Number / Number
20/5 = 4 Arithmetic division
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Operators
Relational Operators
% Modulo- Division
Number%Number
20% 3 = 2 a% b = the remainder of the a/b division
^ Power Number ^ Number
2 ^ 3 = 8 Raise to the given power
Table B-2. Arithmetic Operators (Continued)
Operator Description Usage Examples Notes
Table B-3. Relational Operators
Name Description Usage Examples Description
< Less Than Number < Number
String < String
Date < Date
5 < 7 is assessed as 1 (true).
“abc” < “axx” means 1 (true).
For strings, it performs the lexicographic comparison.
NOTE: To compare numbers and strings, use the str2num or num2str function.
> Greater Than Number > Number
String > String
Date > Date
5 > 7 is assessed as 0 (false).
“John” > “Alex” means 1 (true).
For strings, it performs the lexicographic comparison.
<= Less Or Equal Number <= Number
String <= String
Date <= Date
5 <= 5.2 is evaluated as 1 (true).
“Abc” <= “Abcd” is assessed as 1 (true).
For strings, it performs the lexicographic comparison.
>= Greater Or Equal
Number >= Number
String >= String
Date >= Date
5 >= 5.2 is assessed as 0 (false).
“XYZ” >= “ABC” is assessed as 1 (true).
For strings, it performs the lexicographic comparison.
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Logical Operators
== Equal Number == Number
String == String
Date == Date
5 == 5.2 is assessed as 0 (false).
“XYZ” == “XY” + “Z” is assessed as 1 (true).
For strings, it performs the lexicographic comparison.
!= Different Number != Number
String != String
Date != Dat
5!= 5.2 is assessed as 1 (false).
“XYZ”!= “xy” + “Z” is assessed as 1 (true).
For strings, it performs the lexicographic comparison.
Table B-3. Relational Operators (Continued)
Name Description Usage Examples Description
Table B-4. Logical Operators
Operator Description Usage Examples Notes
&& AND Number && Number
(6 – 5 == 1) && (7 > 2) is assessed as 1 (true).
0 && 0 is assessed as 0.
1 && 0 is assessed as 0.
0 && 1 is assessed as 0.
1 && 1 is assessed as 1.
|| OR Number || Number
(“abc” == “a”) || (4 > 3) is assessed as 1 (true).
0 || 0 is assessed as 0.
1 || 0 is assessed as 1.
0 || 1 is assessed as 1.
1 || 1 is assessed as1.
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Operators
Format Operator Format – Configures the output format of another variable or string of charac-ters.
The behavior of the Format operator is similar with the output format from the Visual C printf function.
The syntax is: format(“%TypeFieldCharacter”, string).
The available TypeFieldCharacters are described in Table B-5.
! NOT ! Number (unary operator)
! (“abc” == “a”) is assessed as 1 (true).
! 0 is assessed as 1.
! 1 is assessed as 0.
? : Conditional Operator
Number ?
(Number or String or Date) : (Number or String or Date)
($MapPos % 2 == 0) ? “Even” : “Odd”
is equivalent with:
If the channel index is an even number, (remainder upon division by 2 is zero) the expression evaluates as “Even”; otherwise, it evaluates as “Odd.”
Condition? Expression1: Expression2.
If the condition evaluates as true, then Expression1 is evaluated; otherwise, Expression2 is evaluated.
NOTE: For all logical operations, 1 means true and 0 means false.
Table B-4. Logical Operators (Continued)
Operator Description Usage Examples Notes
NOTE: The Format operator returns the character associated to the ASCII code.
Table B-5. Format Operator
Character Type Output Format
C Int Character
C Int or Wint_t When used with the printf functions, it specifies a wide character; when used with the wprintf functions, it specifies a single-byte character.
D Int Signed decimal integer
I Int Signed decimal integer
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OperatorsB
O Int Unsigned octal integer
U Int Unsigned decimal integer
X Int Unsigned hexadecimal integer using “abcdef”
X Int Unsigned hexadecimal integer using “ABCDEF”
E Double Signed value having the [-]d.dddd e [sign]ddd form, where d is a single decimal digit, dddd is one or more decimal digits, ddd is exactly three decimal digits, and the sign is + or -.
E Double Identical to the e format, except that E rather than e introduces the exponent.
F Double Signed value having the [-]dddd.dddd form, where dddd is one or more decimal digits. The number of digits before the decimal point depends on the magnitude of the number, and the number of digits after the decimal point depends on the requested precision.
G Double Signed value printed in f or e format, whichever is more compact for the given value and precision. The e format is used only when the exponent of the value is less than -4 or greater than or equal to the precision argument. Trailing zeros are truncated, and the decimal point appears only if one or more digits follow.
G Double Identical to the g format, except that E, rather than e, introduces the exponent (where appropriate).
N Pointer To Integer
Number of characters successfully written so far to the stream or buffer; this value is stored in the integer whose address is given as the argument.
P Pointer To Void
Prints the address pointed to by the argument in the xxxx:yyyy format, where xxxx is the segment and yyyy is the offset, and the x and y digits are the uppercase hexadecimal digits.
S String When used with printf functions, it specifies a single-byte character string; when used with wprintf functions, it specifies a wide-character string. Characters are printed up to the first null character or until the precision value is reached.
S String When used with printf functions, it specifies a wide-character string; when used with wprintf functions, it specifies a single-byte-character string. Characters are printed up to the first null character or until the precision value is reached.
Table B-5. Format Operator (Continued)
Character Type Output Format
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Operators
Functions Supported functions are listed in Table B-6:
Table B-6. Functions
Function
Description Usage Examples
str2num Converts a string to a number. The string should represent a real number (you cannot convert “abc” to a number, but you can convert “100” to the number 100).
str2num(string) str2num(“3”+“5”) = 35
num2str Converts a number to a string.
num2str(number) num2str(3 + 5) = “8”
length Calculates a string length. length(string) Set Variable $string_var == "abc"
lenght($string_var) = 3
extract Extracts a text substring from a string.
extract(string,start_ position, end_ position)
Set Variable $string_var == "abcdefgh"
extract($string_var, 2,5) = "bcde"
lookup Searches for a a text in a string and returns 1 (true) if the text is found and 0 (false) otherwise.
lookup(string, text) lookup($SIP_Message,"TO:")
tick Stores the time elapsed from the last restart of the machine in a user-defined variable. It is used for time difference measurements.
tick(void) tick()
hrtick Stores the time elapsed from the last restart of the machine in a user-defined variable. It is used for time difference measurements with high resolution. As compared to tick, it has a 12 millisecond (ms) resolution
hrtick(void) hrtick()
random Returns a random number in the specified range.
random(min_number,max_number)
random(31,75) =54
generateguid
Returns a globally unique identifier (GUID) with a specified prefix and length. If no length is specified, the endpoint’s MAC address string is appendex to the prefix.
generateguid (prefix, guid_len)
generateguid ("qwe12",13)
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OperatorsB
Compound Variables
A compound variable contains a number of internal fields.
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CAppendix C: Using the H248 Descriptor Editor
The following sections help you configure descriptors for H.248/MEGACO commands using the Descriptor Editor.
The Descriptor Editor GUI
An H.248 descriptor is en entity within a command carrying parameters related to a specific function of the protocol.
For each H.248 command, the Descriptors pane displays a list of supported descriptors. When a descriptor is selected in the list, it becomes the currently selected descriptor and its structure is displayed in the Descriptor Editor using a tree representation (Figure C-1).
Figure C-1. Descriptor Editor GUI
The tree structure can contain both non-leaf and leaf (terminal) items.
Depending on the node selected at the tree level, different options are enabled or disabled. For non-leaf items, a list with subcomponents is displayed, while for leaf items an edit field becomes available enabling you to change the value.
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The Descriptor Editor GUIC
Editing Transmitted Request Messages
For a transmitted request message, for each contained message descriptor its descriptor tree is edited for both non-leaf and leaf nodes, as described below.
Non-Leaf Items
A non-leaf item can be of the following types:
• Sequence – This is a collection of sub-items, whereby each sub-item may be mandatory or optional. For an item of type sequence, if the item has optional sub-items, these will have a checkbox attached in the list with sub-components to enable/disable them.
An example is the localControlDescriptor in the mediaDescriptor.
• Sequenceof – This a set of subitems of the same type.
(e.g. propertyParms in localControlDescriptor)
• Choice – This a single sub-item that is selected from a subitems set.
To edit a non-leaf item:
1. Select a node in the tree representation and activate optional subitems by right-clicking the node (Figure C-2). A context menu displays the list of the node’s subitems.
2. For an item of type sequence, click to check/uncheck the subitems in the Name pane at the right. The selected subitems are added to the tree list and the Name section updates to reflect the selection status.
Figure C-2. Descriptors - Context Menu
For an item of type choice, the list with subitems has the possible choices under the Name section. Same choices are also available on the popup menu by right-clicking on the tree item (Figure C-3).
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The Descriptor Editor GUI
Figure C-3. Descriptors - Item Type Choice
For an item of type sequenceof, all existing subitems are available under the sub-component list with a check option. The last entry is unchecked and used to add new subitems. Unchecking a sub-item deletes it (Figure C-4).
Figure C-4. Descriptors - Item Type Sequenceof
Leaf Items
Leaf items may be of type string , number , boolean , or enum . The following options are available:
• Auto - If selected, this option sets an activity-level automatic value for the parameter at run-time. This option is available only for some elements (for example, RequestID in the Events or localDescriptor – SDP in the Media descriptor).
• Value - Enable this option to specify a value for the message element. When choosing this option, an edit box becomes available below the radio buttons (Figure C-5).
• Expression - Enable this option to specify a run-time evaluated expression for the message element. The expression creation rules are similar with those for expressions in other script functions, for example, a simple variable can be referred to as $variable_name, a global variable can be used as $array_var[4].
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The Descriptor Editor GUIC
When choosing this option, an edit box becomes available below the radio buttons (Figure C-5).
Figure C-5. Descriptors - Leaf Items
For package components, a value is edited as package name and event/signal/property name. Each name is edited in its own combo box., each combo boxes having a list of names from pre-loaded packages (Figure C-6).
Figure C-6. Descriptors - Package Components
For SDP items, click on the Edit SDF button to access the Custom SDP window (Figure C-7).
Figure C-7. Descriptors - SDP Leaf Item
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The Descriptor Editor GUI
In the Custom SDP Editor window, the SDP string can be edited. Several tem-plates are also available. To select a template, click Create from Template drop-down (Figure C-8).
Figure C-8. Descriptors - Custom SDP Editor
The values for leaf items may be saved into user-defined test scenario variables, such as for example for referencing them in other script functions.
To save a sub-item value, select the Save in variable option and choose the name of the scenario variable in the dropdown below (Figure C-9).
Figure C-9. Descriptors - Saving Leaf Items
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The Descriptor Editor GUIC
Editing Expected Response Messages
This operation is largely similar with the editing of transmitted H.248 request messages, in the sense that expected message elements are also specified using the descriptors tree representation shown in Descriptor Editor GUI on page C-1.
An expected message item can be specified using one of the following options:
• Auto - If selected, this option sets an activity-level automatic value for the parameter at run-time. This option is available only for some elements (for example, terminationName or contextID).
• Match Value - If selected, the expected message element must have a specific value that is specified in the pane below.
• Match Expression - If selected, the expected message element is specified using a run-time evaluated expression.
The expression creation rules are similar with those for expressions in other script functions, for example, a simple variable can be referred to as $variable_name, a global variable can be used as $array_var[4].
• Match any value: If selected, the expected message element can have any value.
• Custom SDP string: If selected, the expected SDP definition needs specified in the Custom SDP Editor window that is accessed by clicking the Edit SDP button below.
• Not allowed: If selected, the expected message element cannot be present in the expected reply.
Note: While in the Descriptor Editor, a legend of all node representations can be
obtained by clicking the button, which opens a window such as the following:
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DAppendix D: Using the MGCP Parameter Editor
The following sections provide information for configuring MGCP script func-tion.
MGCP Script Functions Overview
IxLoad MGCP script functions fall into the following categories:
• Send type: These functions implement transactions that consist in sending an MGCP command followed by the receiving of an awaited response message. Such a function sends a command and then waits for any of the specified response messages.
• Wait type: These functions implement MGCP transactions that consist in the receiving of the specified command followed by the sending of a configured response message. Such a function waits for a matching MGCP command with matching parameters and then sends the configured response message.
Since both function types implement a command/response transaction model, for each script function two different configuration pages are available, one for com-mands and another one for responses.
For example, in the case of the Send NTFY script function shown in Figure D-1, the TX Command page is used for specifying the Notify command’s parameters, while in the RX Response page you specify the response message an MGW entity waits for after having sent an NTFY command.
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MGCP Script Functions OverviewD
Figure D-1. MGCP Editor - TX and RX Pages
Script functions can be edited by manually editing their parameters, or by import-ing the commands from a text file.
Functions can also be created starting from templates.
Send Type Functions
Send-type script functions contain a page for specifying the parameters of the sent MGCP command (Tx Command Page) and another one for specifying the expected response message (Rx Response Page).
Tx Command Page
This page is used for editing the parameters of the sent MGCP command.
Any such page contains a command-specific list of mandatory and optional parameters. In Figure D-2 below that shows the Send NTFY command in the Table viewing mode of the editor, the first two parameters (X, O) are mandatory,
while the subsequent ones, N and K – prefixed by the selection control – are optional.
Note: Commands that create or modify connections – CRCX, MDCX – also require you to specify an SDP definintion that represents an endpoint’s media capabilities. Specifyind an SDP is done by either choosing the use of the activity-level settings, or by manually editing the SDP in a separate SDP editor window.
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MGCP Script Functions Overview
Figure D-2. Tx Command Page - Table View
When choosing the Text viewing mode (Figure D-3), the editor displays the com-mand followed by the mandatory parameters only.
Figure D-3. Tx Command Page - Text View
Editing a command parameter is done by clicking the corresponding entry in the Value column and selecting the desired value – a string or an IxLoad VoIP vari-able – instead of the AUTO value that is configured by default.
Rx Response Page
Editing a response is done in the Rx Response / Tx Response page, depending on whether the script function is of the Sent or the Wait type.
By default, the response page is initially configured using a 200 OK response (Figure D-4).
Note: While in the Text viewing modes, you can display/hide special characters by selecting/unselecting the Show / hide CRLF option.
Note: The Auto value specifies an automatic activity-level value for the parameter that is assigned at run-time.
Note: Both editor views support the use of script variables and mathematic expressions comprising variables. Whenever variables are used, these need to be enclosed in "<" and ">", such as for example in <$CallAgentName>.
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MGCP Script Functions OverviewD
Figure D-4. Rx / Tx Response Page
You can add more awaited responses by clicking the Add button and choosing a response from the window that appears. When another response is added to the list of responses to be matched, an additional output is added to the script func-tion.
To remove a response from the list, click the Delete button.
Wait Type Functions Wait-type script functions contain a page for specifying the expected MGCP command and parameters (Tx Command Page) and another one for specifying the response message t(Rx Response Page) hat is sent when a matching command is received.
Rx Command Page
This page, used for specifying the parameters of the expected MGCP command., contains a command-specific list of mandatory and optional parameters, similar to that described in the Tx Command Page.
In addition to the command parameter to be matched, script functions that imple-ment media-related functionality – Wait CRCX and Wait MDCX – contain the following additional options:
• Ignore SDP: If selected, the SDP definition contained in the incoming MGCP command is ignored.
• Extract SDP: If selected, the SDP definition contained in the incoming MGCP command is processed and media capability information is extracted.
Tx Response Page
When an MGCP command matching the definition from the Rx Command page is received, this page is used for specifying an MGCP response message.
By default, the sent response message is an automatically generated message based on the type of the received command and the current state of the MGCP entity receiving the message (Auto response option selected). Alternatively, when de-selecting the Auto response option, you can specify the sent response
Note: For example, assuming the Send NTFY function were configured in the RX Response tab using the 200 OK and the 202 Accepted responses, after sending a NOTIFY command, the MGW would wait for either a 200 OK or a 202 Accepted response message.
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MGCP Script Functions Overview
message by selecting it from a list and edit its parameters, as described in the Tx Command Page section.
In addition to the command parameter to be matched, for script functions that implement media - related functionality – Wait CRCX and Wait MDCX – by default the response also contains an SDP definition with activity-level settings (Use activity settings option is selected).
Alternatively you can edit the sent SDP definition by choosing the Custom SDP
option, clicking the button and editing the definition manually in the editor window that appears.
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MGCP Script Functions OverviewD
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EAppendix E: Skinny Sample Configurations Overview
Table E-1 provides an overview of configuration parameters for the predefined Skinny test samples.
Table E-1. Device Names, Phone Types and Numbers Associated with Skinny Sample Test Configurations
Range of Device Name Range of Phone Numbers
Min # of unique phones required
Phone Type
Group of Test Cases
Feature Settings
SK_001_7902_SO_US_100_Chs_IPv4_Static_Seq_Registration_5_retries
SEP7902A00[00001-] 19[00001-] 100 7902 Bulk Registration
SK_002_7960_SO_US_100_Chs_IPv4_Static_Seq_Registration_5_retries
SEP7960A00[00001-] 16[00001-] 100 7960 Bulk Registration
SK_005_7902_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop
SEP7902A00[00001-] 19[00001-] 100 7902 Bulk Registration
SK_006_7960_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop
SEP7960A00[00001-] 16[00001-] 100 7960 Bulk Registration
SK_003_7902_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop_5_retries
SEP7902A00[00001-] 19[00001-] 100 7902 Bulk Registration
SK_004_7960_SO_US_100_Chs_IPv4_Static_Seq_Bulk_Registration_loop_5_retries
SEP7960A00[00001-] 16[00001-] 100 7960 Bulk Registration
SK_007_7902_SO_US_100_Chs_IPv4_Static_Seq_Reg_5s_Sleep_Dereg_5_retries
SEP7902A00[00001-] 19[00001-] 100 7902 Bulk Registration
SK_008_7960_SO_US_100_Chs_IPv4_Static_Seq_Reg_5s_Sleep_Dereg_5_retries
SEP7960A00[00001-] 16[00001-] 100 7960 Bulk Registration
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E
SK_009_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_10s, SK_010_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_3min, SK_011_7902_SO_US_5000_Chs_IPv4_Static_Basic_Call_30min
SEP7902A00[00001-] 19[00001-] 5000 7902 Bulk Call Testing
SEP7902B00[00001-] 31[00001-] 5000 7902 Bulk Call Testing
SK_012_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_10s, SK_013_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_3min, SK_014_7902_SM_US_900_Chs_IPv4_Static_Basic_Call_Voice_30min
SEP7902A00[00001-] 19[00001-] 900 7902 Bulk Call Testing
SEP7902B00[00001-] 31[00001-] 900 7902 Bulk Call Testing
SK_016_7902_SM_US_300_Chs_IPv4_Static_Basic_Call_DTMFs_inband_3minSK_017_7902_SM_US_300_Chs_IPv4_Static_Basic_Call_DTMFs_out-of-band_3min
SEP7902A00[00001-] 19[00001-] 300 7902 Bulk Call Testing
SEP7902B00[00001-] 31[00001-] 300 7902 Bulk Call Testing
SK_019_7902_SM_US_300_Chs_IPv4_Static_BasicCall_TONE_lowFreq_inband_3min SK_020_7902_SM_US_300_Chs_IPv4_Static_BasicCall_TONE_medFreq_inband_3min SK_021_7902_SM_US_300_Chs_IPv4_Static_BasicCall_TONE_highFreq_inband_3min
SEP7902A00[00001-] 19[00001-] 5000 7902 Bulk Call Testing
SEP7902B00[00001-] 31[00001-] 5000 7902 Bulk Call Testing
SK_015_7902_SM_US_10K_BHCA_IPv4_Static_Basic_Call_Voice_1min
SEP7902A00[00001-] 19[00001-] 125 7902 Bulk Call Testing
SEP7902B00[00001-] 31[00001-] 125 7902 Bulk Call Testing
SK_018_7902_SM_US_25K_BHCA_IPv4_Static_Basic_Call_DTMFs_inband_3_min
SEP7902A00[00001-] 19[00001-] 900 7902 Bulk Call Testing
SEP7902B00[00001-] 31[00001-] 900 7902 Bulk Call Testing
SK_022_7960_SM_US_5_Chs_IPv4_Static_Hold_Resume
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SK_031_7960_SM_US_5_Chs_IPv4_Static_List_Ad_Hoc_Conference
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SEP7960CF0[00001-] 38[00001-] 5 7960 Call Features
SK_032_7960_SM_US_5_Chs_IPv4_Static_Forward_All_Calls
Table E-1. Device Names, Phone Types and Numbers Associated with Skinny Sample Test Configurations
Range of Device Name Range of Phone Numbers
Min # of unique phones required
Phone Type
Group of Test Cases
Feature Settings
E-2 IxLoad Voice Test Library Reference Guide,
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SEP7960CF0[00001-] 38[00001-] 5 7960 Call Features
SK_033_7960_SM_US_5_Chs_IPv4_Static_Forward_Busy
SEP7960AF4[00001-] 48[00001-] 5 7960 Call Features
SEP7960BF4[00001-] 49[00001-] 5 7960 Call Features Forward Busy
Destination D (51[00001-]),Busy Trigger 1
SEP7960CF4[00001-] 50[00001-] 5 7960 Call Features
SEP7960DF4[00001-] 51[00001-] 5 7960 Call Features
SK_034_7960_SM_US_5_Chs_IPv4_Static_Forward_No_Answer
SEP7960AF5[00001-] 52[00001-] 5 7960 Call Features
SEP7960BF5[00001-] 53[00001-] 5 7960 Call Features Forward No Answer
Destination C (54[00001-])
SEP7960CF5[00001-] 54[00001-] 5 7960 Call Features
SK_026_7960_SM_US_5_Chs_IPv4_Static_Ad_hoc_Conference
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SEP7960CF0[00001-] 38[00001-] 5 7960 Call Features
SK_027_7960_SM_US_5_Chs_IPv4_Static_MeetMe_Conference
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SEP7960CF0[00001-] 38[00001-] 5 7960 Call Features
SK_028_7960_SM_US_5_Chs_IPv4_Static_Join_2_Calls
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SEP7960CF0[00001-] 38[00001-] 5 7960 Call Features
SK_024_7960_SM_US_5_Chs_IPv4_Static_Blind_Transfer
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
Table E-1. Device Names, Phone Types and Numbers Associated with Skinny Sample Test Configurations
Range of Device Name Range of Phone Numbers
Min # of unique phones required
Phone Type
Group of Test Cases
Feature Settings
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SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SEP7960CF0[00001-] 38[00001-] 5 7960 Call Features
SK_023_7960_SM_US_5_Chs_IPv4_Static_Transfer_with_Consultation
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SEP7960CF0[00001-] 38[00001-] 5 7960 Call Features
SK_025_7960_SM_US_5_Chs_IPv4_Static_Direct_Transfer_of_2_parties_on_a_line
SEP7960AF0[00001-] 36[00001-] 5 7960 Call Features
SEP7960BF0[00001-] 37[00001-] 5 7960 Call Features
SEP7960CF0[00001-] 38[00001-] 5 7960 Call Features
SK_036_7960_SM_US_5_Chs_IPv4_Static_Call_Group_Pickup
SEP7960AF1[00001-] 39[00001-] 5 7960 Call Features Call Group A (1330)
SEP7960BF1[00001-] 40[00001-] 5 7960 Call Features Call Group B (1330)
SEP7960CF1[00001-] 41[00001-] 5 7960 Call Features Call Group C (1331)
SK_037_7960_SM_US_5_Chs_IPv4_Static_Call_Pickup
SEP7960AF2[00001-] 42[00001-] 5 7960 Call Features Call Group A (1330)
SEP7960BF2[00001-] 43[00001-] 5 7960 Call Features Call Group B (1330)
SEP7960CF2[00001-] 44[00001-] 5 7960 Call Features Call Group C (1330)
MIX_023_7960_SO_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_10s MIX_024_7960_SO_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_3min MIX_025_7960_SO_US_3000_Chs_IPv4_Static_SK_to_SIP_Call_30_min
SEP7960A00[00001-] 16[00001-] 3000 7960 Mixed SIP -Skinny, SIP UAs
Note: SIP endpoints emulated by the VoIPSIPPeer test activity use registration names and phone numbers defined by the 7960BBBB[0000-] and 30[00001-] sequence generating expressions.
MIX_026_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_Sk_Call_10s MIX_027_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_Sk_Call_3min MIX_028_7960_SO_US_3000_Chs_IPv4_Static_SIP_to_Sk_Call_30_min
SEP7960A00[00001-] 16[00001-] 3000 7960 Mixed SIP -Skinny, SIP UAs
Note: SIP endpoints emulated by the VoIPSIPPeer test activity use registration names and phone numbers defined by the 7960BBBB[0000-] and 30[00001-] sequence generating expressions.
Table E-1. Device Names, Phone Types and Numbers Associated with Skinny Sample Test Configurations
Range of Device Name Range of Phone Numbers
Min # of unique phones required
Phone Type
Group of Test Cases
Feature Settings
E-4 IxLoad Voice Test Library Reference Guide,
MIX_020_7960_SM_US_900_Chs_IPv4_Static_SIP_to_Sk_Call_Voice_10s MIX_021_7960_SM_US_900_Chs_IPv4_Static_SIP_to_Sk_Call_Voice_3min MIX_022_7960_SM_US_900_Chs_IPv4_Static_SIP_to_Sk_Call_Voice_30min
SEP7960A00[00001-] 16[00001-] 900 7960 Mixed SIP -Skinny, SIP UAs
Note: SIP endpoints emulated by the VoIPSIPPeer test activity use registration names and phone numbers defined by the 7960BBBB[0000-] and 30[00001-] sequence generating expressions.
MIX_017_7960_SM_US_900_Chs_IPv4_Static_Sk_to_SIP_Call_Voice_10s MIX_018_7960_SM_US_900_Chs_IPv4_Static_Sk_to_SIP_Call_Voice_3min MIX_019_7960_SM_US_900_Chs_IPv4_Static_Sk_to_SIP_Call_Voice_30min
SEP7960A00[00001-] 16[00001-] 900 7960 Mixed SIP -Skinny, SIP UAs
Note: SIP endpoints emulated by the VoIPSIPPeer test activity use registration names and phone numbers defined by the 7960BBBB[0000-] and 30[00001-] sequence generating expressions.
MIX_031_7960_SO_US_75k_BHCA_IPv4_Static_Sk_to_SIP_Call MIX_029_7960_SM_US_75k_BHCA_IPv4_Static_Sk_to_SIP_Call_Voice
SEP7960A00[00001-] 16[00001-] 3000/900 7960 Mixed SIP -Skinny, SIP UAs
Note: SIP endpoints emulated by the VoIPSIPPeer test activity use registration names and phone numbers defined by the 7960BBBB[0000-] and 30[00001-] sequence generating expressions.
MIX_032_7960_SO_US_75k_BHCA_IPv4_Static_SIP_to_Sk_Call MIX_030_7960_SM_US_75k_BHCA_IPv4_Static_SIP_to_Sk_Call_Voice
SEP7960A00[00001-] 16[00001-] 3000/900 7960 Mixed SIP -Skinny, SIP UAs
Note: SIP endpoints emulated by the VoIPSIPPeer test activity use registration names and phone numbers defined by the 7960BBBB[0000-] and 30[00001-] sequence generating expressions.
MIX_001_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_10s MIX_002_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_3min MIX_003_7960_SO_US_5k_Chs_IPv4_Static_SK_to_SIP_trunk_Bulk_Call_30min
SEP7960A00[00001-] 16[00001-] 5000 7960 Mixed SIP -Skinny, SIP Trunk
Note: The SIP endpoints emulated by the VoIPSIPPeer test activity use phone numbers defined by the 20000[00001-] sequence generating expression.
MIX_004_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_10s MIX_005_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_3min MIX_006_7960_SO_US_5k_Chs_IPv4_Static_SIP_to_SK_trunk_Bulk_Call_30min
SEP7960A00[00001-] 16[00001-] 5000 7960 Mixed SIP -Skinny, SIP Trunk
Table E-1. Device Names, Phone Types and Numbers Associated with Skinny Sample Test Configurations
Range of Device Name Range of Phone Numbers
Min # of unique phones required
Phone Type
Group of Test Cases
Feature Settings
IxLoad Voice Test Library Reference Guide, Release 8.00 E-5
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Note: The SIP endpoints emulated by the VoIPSIPPeer test activity use phone numbers defined by the 20000[00001-] sequence generating expression.
MIX_012_7960_SM_US_900_Chs_IPv4_Static_SIP_SK_trunk_Call_Voice_10s MIX_013_7960_SM_US_900_Chs_IPv4_Static_SIP_SK_trunk_Call_Voice_3min MIX_014_7960_SM_US_900_Chs_IPv4_Static_SIP_SK_trunk_Call_Voice_30min
SEP7960A00[00001-] 16[00001-] 900 7960 Mixed SIP -Skinny, SIP Trunk
Note: The SIP endpoints emulated by the VoIPSIPPeer test activity use phone numbers defined by the 20000[00001-] sequence generating expression.
MIX_009_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunk_Call_Voice_10s MIX_010_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunk_Call_Voice_3min MIX_011_7960_SM_US_900_Chs_IPv4_Static_SK_to_SIP_trunk_Call_Voice_30min
SEP7960A00[00001-] 16[00001-] 900 7960 Mixed SIP -Skinny, SIP Trunk
Note: The SIP endpoints emulated by the VoIPSIPPeer test activity use phone numbers defined by the 20000[00001-] sequence generating expression.
MIX_008_7960_SO_US_80k_BHCA_IPv4_Static_SIP_to_SK_trunk_Bulk_Call
SEP7960A00[00001-] 16[00001-] 5000 /900 7960 Mixed SIP -Skinny, SIP Trunk
Note: The SIP endpoints emulated by the VoIPSIPPeer test activity use phone numbers defined by the 20000[00001-] sequence generating expression.
MIX_016_7960_SM_US_75k_BHCA_IPv4_Static_SIP_to_SK_trunk_Call_Voice
SEP7960A00[00001-] 16[00001-] 5000 /900 7960 Mixed SIP -Skinny, SIP Trunk
Note: The SIP trunk endpoints by the VoIPSIPPeer test activity use phone numbers defined by the 20000[00001-] sequence generating expression.
MIX_007_7960_SO_US_80k_BHCA_IPv4_Static_SK_to_SIP_trunk_Bulk_Call
SEP7960A00[00001-] 16[00001-] 5000 /900 7960 Mixed SIP -Skinny, SIP Trunk
Note: The SIP endpoints emulated by the VoIPSIPPeer test activity use phone numbers defined by the 20000[00001-] sequence generating expression.
MIX_015_7960_SM_US_75k_BHCA_IPv4_Static_SK_to_SIP_trunk_Call_Voice
SEP7960A00[00001-] 16[00001-] 5000 /900 7960 Mixed SIP -Skinny, SIP Trunk
Note: The SIP endpoints emulated by the VoIPSIPPeer test activity use phone numbers defined by the 20000[00001-] sequence generating expression.
Table E-1. Device Names, Phone Types and Numbers Associated with Skinny Sample Test Configurations
Range of Device Name Range of Phone Numbers
Min # of unique phones required
Phone Type
Group of Test Cases
Feature Settings
E-6 IxLoad Voice Test Library Reference Guide,
EAppendix E: Support for Multipart SIP Messages
The IxLoad Voice Plug-in supports SIP messages having multipart bodies, whereby each part is encoded using the MIME format. In addition to the most common application/sdp type, the Content-Type parameter supports the following multipart extensions:
• multipart/mixed: This extension is used for sending additional information to that contained in the SDP. The additional information can be an UE location, an XML text containing resources, and images.
• multipart/alternative: This extension is used for sending different versions (formats) of the same information.
• multipart/related: This extension is used for sending an object comprised of multiple related elements, for example a web page containing multiple images.
Whenever a multipart message body is created, the different parts have to be sep-arated by the boundary parameter, and the body must also be terminated using this separator string. For example, assuming we had a SIP message that contains both an sdp and an xml part, the Content-Type and the separator definition would be as shown in the following example:
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E-2 IxLoad Voice Test Library Reference Guide, Release 8.00